Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 12:07 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Mon, Jun 16, 2008 at 11:11:00AM -0400, Steve Totaro wrote: On Mon, Jun 16, 2008 at 10:35 AM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sun, Jun 15, 2008 at 01:25:18PM -0400, Alex Balashov wrote: Is there a contradiction between them? Naw; Steve's just showin' his ass again. Nah, just showing various tactics, sure some contradict each other. Yes, but clearly, neither Alex nor I thought that the two you quoted actually *do* contradict one another. He was just being polite. Cheers, -- jra -- That will be the day, Alex and Jay being polite. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis It seems you get these goodies at Astricon events. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
I was looking for an option to weigh against a full blown asterisk system. If I use asterisk as an expensive ata then there isn't much point in keeping the key system is there? While I could hack a solution together (pap2 and Wrt54gs running * on openwrt comes to mind) I'd really rather not. Unless I do a full blown * box for the client I'm looking for a simple ata solution. Eric On Mon, Jun 16, 2008 at 10:37 PM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Mon, 16 Jun 2008, Eric Fort wrote: I'm presently working on provisioning VoIP to a traditional key system. I have a single SIP DID inbound that gives me a maximum of 2 concurrent channels. I need an ATA that will ring the second station port when the first is in use. What devices will do this with a single sip registration with the provider? Er, Asterisk will do this with a TDM400 card or clone. Expensive ATA though :) But maybe an AVM Fritz! box will work for you too... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 10:03 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis It seems you get these goodies at Astricon events. Unfortuneately it's too far and too expensive for me to get there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg call recording
On Tue, Jun 17, 2008 at 8:34 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Bikrish Amatya wrote:
Hi all
I am using asterisk as pbx for my company. My company has requirement
that all the incoming and outgoing calls should be recorded for all the
extensions and should be able to play recorded call on extensions basis,
that is , say 123 extension has made what call on the particular date
and should be able to play and listen to it. What is the better way to
achieve this? Any kind of suggestion is truly appreciated.
Bikrish
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
A simple web interface, such as asterisk-stats coupled with some basic
modifications to link to a recording that was made with ${UNIQUEID} as
the recording filename (pre extension, use monitor + soxmix to mix the
recordings) will work just fine, I use it on a medium-large installation
that does about 10K calls a day, with no issues in regards to recordings
or ability to access calls/recordings.
I have similar setup, and here are some suggestions from my experience.
Do recording only in native format, that will decrease the load by
transcoding at working time.
Whenever somebody requests to listen, you can mix, transcode and play.
This usually takes few seconds (however depends on call duration). Mix
and transcode (to some lower bandwidth codec) the rest of recordings
at night time.
Personally I record everything in ulaw, and either on listen or at
night transcode to gsm for storage.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Dear Sherwood,
Thanks.
Just three questions:
1. Will I be needing Apache or Asterk-stat will handle itself?
2. Are there How-tos for integerating asterisk-stat with asterisk?
3. My Recordings are being saved in the default folder i.e:
/var/spool/asterisk/monitor/ in .gsm format. When I wish to listen to a
particular recording I first convert it with SOX utility into .wav
format and then listen it. Will this also be automated so that when I
select a recording and try to listen it will be in right format.
Thanks again.
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
Syed Nasruddin wrote:
Dear Sherwood,
I am also using Asterisk Call Center Setup in my office with voice
recording. The only thing I am unable to setup is web based call
recording (CDR) access. From your email I think you have configured
such
a thing can you please share with me how can I also setup this
solution.
I know how to run and install Apache. Don't know abt PostgreSQL.
However
can do it if you can define some steps.
And also how to integrate this all PostgreSQl+Apache+Web Based Links
to
Recordings. It will be a great help.
regards
Syed Nasruddin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sherwood
McGowan
Sent: Tuesday, June 17, 2008 5:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Center
broadband Voice wrote:
Is anyone using Asterisk as a call center. I want to be able to set
it
up for my office line, when calls come in after 7:00pm Est want a
recording to says the office is closed and have about 5 phones that I
want to use as an agent. Can anyone share their implementation?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
There's a ton of us on here who have installations in call centers.
What
would you like to know?
I currently have a Dual AMD64 2.4Ghz (Dual Cores on each) with 4GB RAM
running a Tormenta 2 and a Digium 407. Two T1s going to a PRI, 12 FXO
channels in a Rhino modular channel bank (all on the Digium card), and
2
24 port adtran total access channel banks running on the Tormenta. The
Adtrans drive the 40 analog phones for the sales floor, and we have 25
SIP phones. All phone conversations are recording by Asterisk and are
converted from GSM to Speex post-call by speexenc. We also run
PostgreSQL and Apache on the same system to serve CDRs with links to
recordings.
Anything else you'd like to know?
Syed,
What I did for a quick and dirty solution was install asterisk-stats and
modify the source code to include a link to the unique filename of the
recording (I use ${UNIQUEID}). This has worked just fine for our 75 or
so phone setup :) IIRC we found asterisk-stats on voip-info.org. We
just used that instead of creating an in house CDR web app, since the
client just needed a basic interface to look up calls and pull the
recordings.
If you'd like more information just let me know.
--
Sherwood McGowan
VoIP / Telecom Solutions
[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] voicemail problem
Hi: I configured asterisk for voicemail service.My main configuration files are: extensions.conf [from-pstn] exten =gt; 9711315,1,Dial(SIP/3000,30) exten =gt; 9711315,2,VoiceMail([EMAIL PROTECTED]) exten =gt; 9711315,3,PlayBack(vm-goodbye) exten =gt; 9711315,4,HangUp() voicemail.conf [ff_tutorial] 555 =gt; 1234567,3000,[EMAIL PROTECTED] sip.conf [3000] type=friend username=3000 secret=1234567 host=dynamic context=from-pstn [EMAIL PROTECTED] But when I dial 9711315, after 30s I hear goodbye and call hangups. in console: -- Accepting call from '3322000' to '9711315' on channel 0/2, span 1 -- Executing Dial(Zap/2-1, SIP/3000|30) in new stack -- Called 3000 -- SIP/3000-08f18698 is ringing Jun 24 11:55:32 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 24 11:55:42 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 24 11:55:52 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Nobody picked up in 3 ms -- Executing VoiceMail(Zap/2-1, [EMAIL PROTECTED]) in new stack Jun 24 11:55:53 WARNING[5188]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '555' -- Executing Playback(Zap/2-1, vm-goodbye) in new stack -- Playing 'vm-goodbye' (language 'en') -- Executing Hangup(Zap/2-1, ) in new stack == Spawn extension (from-pstn, 9711315, 4) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' Jun 24 11:56:02 WARNING[4112]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 what's problem? should I do something in sip phone for voicemail? I'd appreciate any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Tue, 17 Jun 2008, randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? This might depends on your country (re. availability), but I've had a lot of good results with the Siemens DECT range... (eg. S450IP) The base-station has a built in ATA, so 2 sockets, one PSTN, one Ethernet... Although if you already have a DECT phone, then who knows - I've used Grandstreams and they're OK, bu seem a bit laggy on answering - ie. pickup, then nothing for a second... Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes
Afternoon All, Does anyone here have any experience with an Audiocodes Mediant 2000? I know its a bit 'non asterisk' but i figured you guys are as likely as any to have come across them. I'm having a few problems with one, i.e. its not sending screening/privacy flags although it is sending caller ID field. If anyone has come across this (or is willing to give their 2 cents) i would be really grateful. Thanks in advance. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: OT How Digium Saved My Bacon!
On June 17, 2008 01:45:43 am randulo wrote: The screwdriver is reversible, it swings both ways, pull out the shank and stick it in the other way, it becomes a Phillips. I'm tellin ya, there Digium engineers are good! Most every pocket screwdriver that is sold as a promotional item is like that. It's not always good; I cut my hand pretty badly when the phillips end slid clean through the screwdriver and into my hand once. I wonder if they'll consider a slot/robertson combination for us northerners. :-) -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with realtime?
I get that a lot since moving to 1.4.21 (from 1.4.18 or something). [Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Question 1: what debug file should I be looking at? Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with realtime?
Just an addition: that happens big time when I do a sip reload from the CLI I know this should help me already, but it doesn`t From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Sent: Tuesday, June 17, 2008 09:23 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Problem with realtime? I get that a lot since moving to 1.4.21 (from 1.4.18 or something). [Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Question 1: what debug file should I be looking at? Mick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail problem
On Tuesday 17 June 2008 04:05:58 fateme fatah wrote: I configured asterisk for voicemail service.My main configuration files are: voicemail.conf [ff_tutorial] 555 =gt; 1234567,3000,[EMAIL PROTECTED] But when I dial 9711315, after 30s I hear goodbye and call hangups. in console: -- Accepting call from '3322000' to '9711315' on channel 0/2, span 1 -- Executing Dial(Zap/2-1, SIP/3000|30) in new stack -- Called 3000 -- SIP/3000-08f18698 is ringing -- Nobody picked up in 3 ms -- Executing VoiceMail(Zap/2-1, [EMAIL PROTECTED]) in new stack Jun 24 11:55:53 WARNING[5188]: app_voicemail.c:2461 leave_voicemail: No entry in voicemail config file for '555' Did you reload after changing voicemail.conf? What is the output of 'voicemail show users'? -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg call recording
What I have done for our office is actually built my own interface with php and used our SQL database to store the information. Basically I keep all the recordings in gsm format, and store them however I want. I use MixMonitor and use DeadAGI to run a script to rename the file and move it to the directory for that extension. So in your exmaple, .../recordings/123/[file name] I also used session information from the login page to store the person's extension (which we also have in the DB, but there are other ways to do this) that is looking at the interface so when play want to listen to the call, it will generate a call file and dial their phone and playback the file (works nice if you don't have speakers/headphones). Or they can download it. Downloading it will run a script to convert it to wav. I don't know of a best way to do this. I know if you take the time and put the effort, you can get what you/your company wants if you build your own. Or go with some of the other suggestions made which also work perfectly well. I think for me it took about 2 weeks to fully build/test everything and I was coding it by myself (on top of other responsibilities at work). Kevin Bikrish Amatya wrote: Hi all I am using asterisk as pbx for my company. My company has requirement that all the incoming and outgoing calls should be recorded for all the extensions and should be able to play recorded call on extensions basis, that is , say 123 extension has made what call on the particular date and should be able to play and listen to it. What is the better way to achieve this? Any kind of suggestion is truly appreciated. Bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Kevin Smith --- Mercury Network Technical Support Phone: 989.837.3790 Toll Free: 888.866.4638 www.mercury.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with realtime?
On Tuesday 17 June 2008 08:23:08 Mike wrote: I get that a lot since moving to 1.4.21 (from 1.4.18 or something). [Jun 17 09:19:54] WARNING[22053]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Question 1: what debug file should I be looking at? Check your logger.conf to see if you're even logging debug output someplace. (I'm not quite sure why somebody decided that error message was a good idea: if there was a problem, why not tell people what the problem was, instead of telling them to look elsewhere?) -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
Is it possible on a TE220p to deactivate the hardware echo canceler at will ? (With a function in the dialpan for example) example for fax SDA ,beeing able to shutdown the echo canceler could give better results don't you think ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] looking for help / input with Blind transfer from asterisk to zap
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel - Asterisk - SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension - Asterisk - Zaptel
During this whole process, the original channel off the trunk
(lineside T1) is used for the blind transfer (hookflash)
---
[from-sip]
exten = _NXXX,1,Flash()
exten = _NXXX,n,SendDTMF(${EXTEN})
exten = _NXXX,n,Hangup()
[from-zap]
exten = s,1,Dial(SIP/sv0071iv)
exten = s,n,Dial(SIP/sv0072iv)
exten = s,n,Goto(AA,s,1)
[AA]
exten = s,1,Wait(.5)
exten = s,n,Background(vm-whichbox)
exten = s,n,WaitExten
exten = _5XXX,1,Playback(transfer)
exten = _5XXX,n,Flash()
exten = _5XXX,n,SendDTMF(${EXTEN})
exten = _5XXX,n,Hangup()
---
Now, for whatever reason if sv0071iv, and sv0072iv fail to qualify),
asterisk will play a simple menu choice asking which extension they
want to transfer too (Mind the Background message, this part is not
finished).
Zaptel - Asterisk - Blind Transfer Zaptel (hookflash)
The problem is, I'm having trouble getting asterisk to do the blind
transfer. As you see, I'm using the same logic to hookflash over,
send DTMF and hand up.
Any I missing something?
Thanks again,
PB
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for help / input with Blind transfer from asterisk to zap
I always seem to figure my issues just after I post to the list. Had
to add a Wait(.5) after the hookflash.
--
[AA]
exten = s,1,Wait(.5)
exten = s,n,Background(vm-whichbox)
exten = s,n,WaitExten
exten = _5XXX,1,Playback(transfer)
exten = _5XXX,n,Flash()
exten = _5XXX,n,Wait(.5)
exten = _5XXX,n,SendDTMF(${EXTEN})
exten = _5XXX,n,Hangup()
Thanks again,
PB
On Tue, Jun 17, 2008 at 11:15 AM, Paul Belanger [EMAIL PROTECTED] wrote:
List,
Having a little trouble with the following. Let me prefix by saying I
have blind transfers working from the following setup.
Inbound call [from-zap] (SIP/sv0071iv) answers.
Zaptel - Asterisk - SIP extension
SIP extension then blind transfers [from-sip]
---
SIP extension - Asterisk - Zaptel
During this whole process, the original channel off the trunk
(lineside T1) is used for the blind transfer (hookflash)
---
[from-sip]
exten = _NXXX,1,Flash()
exten = _NXXX,n,SendDTMF(${EXTEN})
exten = _NXXX,n,Hangup()
[from-zap]
exten = s,1,Dial(SIP/sv0071iv)
exten = s,n,Dial(SIP/sv0072iv)
exten = s,n,Goto(AA,s,1)
[AA]
exten = s,1,Wait(.5)
exten = s,n,Background(vm-whichbox)
exten = s,n,WaitExten
exten = _5XXX,1,Playback(transfer)
exten = _5XXX,n,Flash()
exten = _5XXX,n,SendDTMF(${EXTEN})
exten = _5XXX,n,Hangup()
---
Now, for whatever reason if sv0071iv, and sv0072iv fail to qualify),
asterisk will play a simple menu choice asking which extension they
want to transfer too (Mind the Background message, this part is not
finished).
Zaptel - Asterisk - Blind Transfer Zaptel (hookflash)
The problem is, I'm having trouble getting asterisk to do the blind
transfer. As you see, I'm using the same logic to hookflash over,
send DTMF and hand up.
Any I missing something?
Thanks again,
PB
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange SIP-SIP delay
I've got the following setup: PhoneA - router - vpn - router- asterisk (SIP / ISDN) PhoneB - asterisk (SIP / ISDN) If phone A is connected to phone B (sip-sip), there is a noticable delay (up to 2-3 seconds) between me speaking and the other end hearing. If phone A calls out via the ISDN and back in to the DDI of phone B (ie SIP-ISDN-ISDN-SIP) then there is no delay at all ! Any clues on where I might start looking for this ? Julian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange SIP-SIP delay
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I've got the following setup: PhoneA - router - vpn - router- asterisk (SIP / ISDN) PhoneB - asterisk (SIP / ISDN) If phone A is connected to phone B (sip-sip), there is a noticable delay (up to 2-3 seconds) between me speaking and the other end hearing. If phone A calls out via the ISDN and back in to the DDI of phone B (ie SIP-ISDN-ISDN-SIP) then there is no delay at all ! Any clues on where I might start looking for this ? Julian Have you tested the latency across your VPN? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
Benoit Plessis wrote: Is it possible on a TE220p to deactivate the hardware echo canceler at will ? (With a function in the dialpan for example) example for fax SDA ,beeing able to shutdown the echo canceler could give better results don't you think ? All echo cancelers using Zaptel/DAHDI already disable themselves when FAX or modem communications are used, based on reception and detection of the CED tone that FAX machines and modems generate to make that happen. You can tell this happened by looking at the channel in Asterisk using 'zap show channel' or 'dahdi show channel' as it will show you that the echo canceler was disabled automatically. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
LLCs? On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange SIP-SIP delay
Hi Steve - the vpn is a consistent as the sip-IDSN has to go through the VPN first to get to asterisk. i.e. to make an outside call, PhoneA goes through the vpn to the asterisk box, and out through isdn. Julian Steve Totaro wrote: On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I've got the following setup: PhoneA - router - vpn - router- asterisk (SIP / ISDN) PhoneB - asterisk (SIP / ISDN) If phone A is connected to phone B (sip-sip), there is a noticable delay (up to 2-3 seconds) between me speaking and the other end hearing. If phone A calls out via the ISDN and back in to the DDI of phone B (ie SIP-ISDN-ISDN-SIP) then there is no delay at all ! Any clues on where I might start looking for this ? Julian Have you tested the latency across your VPN? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Putting incoming sip call leg on MOH while dialing out other party**********NEED HELP************
Dear All I need help to implement the follwoing Senario: 1- Incoming SIP call comes to asterisk and putting caller on MOH 2- While the caller is on MOH , dialing out other party and when asterisk recive ANSWER , MOH should be disconnected, then bridging the 2 call legs Appreciate your Help Mohammad Mirzaee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
You are probably confusing corporate tactics to pay less taxes vs corporate tactics to protect assets. The first does provide some asset protection but is mainly to pay less taxes. The second is to basically hide assets through totally legal LLCs. Thanks, Steve Totaro On Tue, Jun 17, 2008 at 12:00 PM, C F [EMAIL PROTECTED] wrote: LLCs? On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
Kevin P. Fleming wrote: Benoit Plessis wrote: Is it possible on a TE220p to deactivate the hardware echo canceler at will ? (With a function in the dialpan for example) example for fax SDA ,beeing able to shutdown the echo canceler could give better results don't you think ? All echo cancelers using Zaptel/DAHDI already disable themselves when FAX or modem communications are used, based on reception and detection of the CED tone that FAX machines and modems generate to make that happen. You can tell this happened by looking at the channel in Asterisk using 'zap show channel' or 'dahdi show channel' as it will show you that the echo canceler was disabled automatically. I thought the ec gets disabled only by the ec disable tone and not the CED tone. regards Hans ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Packages for ubuntu
Hi, Did someone try to package new releases for ubuntu version like gutsy/hardy ? thanks -- Cyril SCETBON ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Putting incoming sip call leg on MOH while dialing out other party**********NEED HELP************
You can use the m flag of dial on the incoming sip channel, such as: exten = s,n,dial(Local/[EMAIL PROTECTED]|60|m) Fred Posner On Jun 17, 2008, at 12:04 PM, Mohammad Mirzaee wrote: Dear All I need help to implement the follwoing Senario: 1- Incoming SIP call comes to asterisk and putting caller on MOH 2- While the caller is on MOH , dialing out other party and when asterisk recive ANSWER , MOH should be disconnected, then bridging the 2 call legs Appreciate your Help Mohammad Mirzaee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote: On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... LLCs? No, my assertion was that I believe that 'Steve's assertion that it is illegal and a breach of duty for a corporation's officers to own its real estate and lease it back to the company' may be dependent on whether the company is publicly owned or not. I suspect that there is no breach in the case of a private company, because different fiduciary duties pertain. I'll ask my client who's the ex-president of one of the companies I was talking about. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 1:02 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote: On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... LLCs? No, my assertion was that I believe that 'Steve's assertion that it is illegal and a breach of duty for a corporation's officers to own its real estate and lease it back to the company' may be dependent on whether the company is publicly owned or not. I suspect that there is no breach in the case of a private company, because different fiduciary duties pertain. I'll ask my client who's the ex-president of one of the companies I was talking about. Please don't attribute quotes to me that I did not make or even assert. It was CF and maybe Alex Re-read the thread and stop showing your a**.. Comprehension and retention is key. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 01:05:59PM -0400, Steve Totaro wrote: On Tue, Jun 17, 2008 at 1:02 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 12:00:18PM -0400, C F wrote: On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... LLCs? No, my assertion was that I believe that 'Steve's assertion that it is illegal and a breach of duty for a corporation's officers to own its real estate and lease it back to the company' may be dependent on whether the company is publicly owned or not. I suspect that there is no breach in the case of a private company, because different fiduciary duties pertain. I'll ask my client who's the ex-president of one of the companies I was talking about. Please don't attribute quotes to me that I did not make or even assert. It was CF and maybe Alex I'm entirely sorry, Steve; you're right. I misread it. It was CF who made the assertion I was commenting on. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
The asset protection entities are completely legal. There's nothing wrong ipso facto with doing it. The question is only whether they will succeed in protecting your assets when your assets are actually challenged. It depends on the size and scope of the judgment, the circumstances in which it takes place, the quality and nature of the litigation, and so on. It is the gulf between the theoretical and the de facto that I am attempting to illuminate. In practise, in situations where courts and plaintiffs are most rabidly after your assets (i.e. bankruptcies of various stripes), merely having them owned by entities other than the one being sued may not be enough. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
Michael, I agree. Here we use e1s(which have even more channels). Some clients just don't want to change some if their old infrastructure. Thanks Michael Graves wrote: I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
Some customers are locked into two year contracts. That was the answer I got when adding four POTS lines to a system with four BRIs... Thanks, Steve Totaro On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku [EMAIL PROTECTED] wrote: Michael, I agree. Here we use e1s(which have even more channels). Some clients just don't want to change some if their old infrastructure. Thanks Michael Graves wrote: I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk v1.6 queue() continue after answered call
Hi list, we upgraded to v1.6 and have a problem understanding the queue() behaveour of the v1.6 in queues. we try to set the queue up to not hangup if an agent answeres the call but then hangs up again. we would then like the queue to go on in the dialplan. But the queue does not want to go on and hangs up. :-( we triyed to use the c flag and the timeoutrestart both did not work. How could we set up the queue to go on after a call? Hope anybody can help. thank you Martin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packages for ubuntu
On Tue, Jun 17, 2008 at 06:45:30PM +0200, Cyril SCETBON wrote: Hi, Did someone try to package new releases for ubuntu version like gutsy/hardy ? The Ubuntu packages are based on the Debian ones and basically packaged from the same repository. http://pkg-voip.alioth.debian.org/ You can rebuild the package with svn-buildpackage . Some distributions need the backporting hook scripts. Simply run: ./debian/backports/distroname in the build directory. E.g.: ./debian/backports/gutsy You'll probably need to use the option --svn-ignore-new for svn-buildpackage as this will make some local changes. It should then build. If it doesn't, please report so we can update that backport script. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg recording of calls
Hi all I appreciate the help that you have given me on call recording. I would like to share how i achieve the way i wanted. I used monitor and soxmix for this. First i used monitor to record the calls and made use of system command to create directory of each extension and inside each directory of respecitve extension created a sound file name with soxmix to create sound file in data-time format and i gave ftp access to the person of the recording directory. Now he can go have ftp access and go to each directory access the sound file by data and time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] connectivity with oracle database and astreisk
Hi all In my company there is oracle database which has the information about the client. Now my requirement is... when my clients calls to our company .. they should be able to get information about them when they call to our pbx. I mean how can reterive information from oracle database and play it to clients . Thanks in advance Bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
How can they even set such 1234567890 callerIDs anyway? For example, our inter/intra state calling depends a lot on the callerIDs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: June 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 'Together Everywhere'
http://www.msnbc.msn.com/id/25119259/ Anyone know if this was built using Asterisk? Seems like a perfect vehicle for it's deployment. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (Direct) +1-917-207-3420 (Mobile) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
2008/6/16 Syed Nasruddin [EMAIL PROTECTED]: Thanks for the link. I think I will be using this product. It's very, very good. You can hook it up to MySQL instead of sqlite if needed, just e-mail support. -- http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
And there are people like me who still can't get PRI's for less than $1100/month. (Granted, I doubt I'll ever need a pri for the business I am with now, but I was with an ISP for a long time that still supported dial-up and we had 8 PRI's with a bulk discount that got them for us at $1000/PRI/month. ($8000/month total). Analog lines with incoming service only were priced at roughly $15/month. Steve Totaro wrote: Some customers are locked into two year contracts. That was the answer I got when adding four POTS lines to a system with four BRIs... Thanks, Steve Totaro On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku [EMAIL PROTECTED] wrote: Michael, I agree. Here we use e1s(which have even more channels). Some clients just don't want to change some if their old infrastructure. Thanks Michael Graves wrote: I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
Johann Steinwendtner wrote: I thought the ec gets disabled only by the ec disable tone and not the CED tone. The CED tone *is* the echo canceler disable tone. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
I like Grandstream 286s No, seriously Thanks, Steve T On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Together Everywhere'
Dean One of the firms involved, playareacode.comhas a division called Big Games, and they used Asterisk as a platform to create an interactive mystery game. http://itp.nyu.edu/blogblender/2007/10/15/the-mystery-of-the-beautiful-c igar-girl-location-plotting-and-ia/ I'd place my bets on asterisk being the telephony platform for the promotion you described in your earlier post. Cory J Andrews Director, New Market Initiatives VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, June 17, 2008 2:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 'Together Everywhere' http://www.msnbc.msn.com/id/25119259/ Anyone know if this was built using Asterisk? Seems like a perfect vehicle for it's deployment. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (Direct) +1-917-207-3420 (Mobile) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Together Everywhere'
Here are some other interesting applications they built off Asterisk http://itp.nyu.edu/blogblender/2007/11/15/voice-recognition-with-lumenvo x/ http://www.prophecyboy.com/itp/redial/booty-dialer-update/ http://www.prophecyboy.com/category/itp/redial/ BootyDialerbestideaever Cory J Andrews Director, New Market Initiatives VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Tuesday, June 17, 2008 3:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 'Together Everywhere' Dean One of the firms involved, playareacode.comhas a division called Big Games, and they used Asterisk as a platform to create an interactive mystery game. http://itp.nyu.edu/blogblender/2007/10/15/the-mystery-of-the-beautiful-c igar-girl-location-plotting-and-ia/ I'd place my bets on asterisk being the telephony platform for the promotion you described in your earlier post. Cory J Andrews Director, New Market Initiatives VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Tuesday, June 17, 2008 2:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 'Together Everywhere' http://www.msnbc.msn.com/id/25119259/ Anyone know if this was built using Asterisk? Seems like a perfect vehicle for it's deployment. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (Direct) +1-917-207-3420 (Mobile) +61-2-9016-5642 (Sydney in-dial) http://www.Cognation.net http://www.Cognation.net/profile ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 1:48 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: Most every pocket screwdriver that is sold as a promotional item is like that. It's not always good; I cut my hand pretty badly when the phillips end slid clean through the screwdriver and into my hand once. Some Linux Day screwdriver set I got was totally worthless. The Digium screwdriver really works. Plus it's more stable than asterisk 1.6. Of course, as a friend of mine once said, (referring to love) when it's dark outside that 25 watt bulb in the closet looks really bright, but when the sun comes up, it ain't s**t! IOW, having now arrived at destination and unpacked a few boxes, my electric screwdriver outshines even the Digium tweaker. Still, it was very handy when all the big gins were packed away. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
and being only a single line device how exactly would I get 2 lines out of it? Eric On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro [EMAIL PROTECTED] wrote: I like Grandstream 286s No, seriously Thanks, Steve T On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
Buy two.. On Tue, Jun 17, 2008 at 3:46 PM, Eric Fort [EMAIL PROTECTED] wrote: and being only a single line device how exactly would I get 2 lines out of it? Eric On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro [EMAIL PROTECTED] wrote: I like Grandstream 286s No, seriously Thanks, Steve T On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 3:46 PM, randulo [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 1:48 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: Most every pocket screwdriver that is sold as a promotional item is like that. It's not always good; I cut my hand pretty badly when the phillips end slid clean through the screwdriver and into my hand once. Some Linux Day screwdriver set I got was totally worthless. The Digium screwdriver really works. Plus it's more stable than asterisk 1.6. Of course, as a friend of mine once said, (referring to love) when it's dark outside that 25 watt bulb in the closet looks really bright, but when the sun comes up, it ain't s**t! IOW, having now arrived at destination and unpacked a few boxes, my electric screwdriver outshines even the Digium tweaker. Still, it was very handy when all the big gins were packed away. Digium is so bigtime they should be giving away re-branded (acid etched so it doesn't rub off) leatherman tools to everyone who has ever ordered from them ;-) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Tue, Jun 17, 2008 at 12:32 PM, Gordon Henderson [EMAIL PROTECTED] wrote: This might depends on your country (re. availability), but I've had a lot of good results with the Siemens DECT range... (eg. S450IP) The base-station has a built in ATA, so 2 sockets, one PSTN, one Ethernet... No question, the Siemens DECT SIP phones are great, I do have one, the S675IP, and I love it. I even wrote a (basic non tech) review of it and here's a photo of the RSS screensaver http://x2z.eu/h click on photo if you want to read the review. I love this phone, biut iot's too expensive for the other location where I just want a SIP ATA to connect to an existing cordless. I will take a look at the Linksys though, sounds interesting. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 9:03 AM, Steve Totaro [EMAIL PROTECTED] wrote: It seems you get these goodies at Astricon events. Thanks, Steve T Digium also gives away the best mouse pad ever and I've gotten dozens of these from every trade show. Theirs is the only one my wife and i fight over for custody. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Tue, Jun 17, 2008 at 3:51 PM, randulo [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 12:32 PM, Gordon Henderson [EMAIL PROTECTED] wrote: This might depends on your country (re. availability), but I've had a lot of good results with the Siemens DECT range... (eg. S450IP) The base-station has a built in ATA, so 2 sockets, one PSTN, one Ethernet... No question, the Siemens DECT SIP phones are great, I do have one, the S675IP, and I love it. I even wrote a (basic non tech) review of it and here's a photo of the RSS screensaver http://x2z.eu/h click on photo if you want to read the review. I love this phone, biut iot's too expensive for the other location where I just want a SIP ATA to connect to an existing cordless. I will take a look at the Linksys though, sounds interesting. DECT can make you and others around you feel very ill. No long term research exists but if immediate effects are feeling ill, it may possibly lead to long term effects. I know a consultant that could barely finish a DECT install because his head hurt so badly. http://www.healthy-house.co.uk/news/2008/01/14/Have-you-or-your-neighbours-got-a-DECT-phone---Benefit-from-our-special-offer-and-test-your-home.php Maybe not the best source to quote but I trust anyone interested can google further. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
They're real cheap where you could get 2 of them. I got one and actually have no complaints. Call quality was really good and it's very, very small and portable. Looks cheap, but you get over that. I have a SPA2102 2 liner which works fine but gets really hot. Fred Posner www.teamforrest.com FWD#: 902963 On Jun 17, 2008, at 3:46 PM, Eric Fort wrote: and being only a single line device how exactly would I get 2 lines out of it? Eric On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro [EMAIL PROTECTED] wrote: I like Grandstream 286s No, seriously Thanks, Steve T On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api- digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
How do 2 of them register only once? -Eric On Tue, Jun 17, 2008 at 12:52 PM, Steve Totaro [EMAIL PROTECTED] wrote: Buy two.. On Tue, Jun 17, 2008 at 3:46 PM, Eric Fort [EMAIL PROTECTED] wrote: and being only a single line device how exactly would I get 2 lines out of it? Eric On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro [EMAIL PROTECTED] wrote: I like Grandstream 286s No, seriously Thanks, Steve T On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 3:53 PM, randulo [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 9:03 AM, Steve Totaro [EMAIL PROTECTED] wrote: It seems you get these goodies at Astricon events. Thanks, Steve T Digium also gives away the best mouse pad ever and I've gotten dozens of these from every trade show. Theirs is the only one my wife and i fight over for custody. I have a You're Fired! mousepad from the Donald. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Tue, Jun 17, 2008 at 9:56 PM, Steve Totaro [EMAIL PROTECTED] wrote: DECT can make you and others around you feel very ill. No long term research exists but if immediate effects are feeling ill, it may possibly lead to long term effects. That may or may not be true, but if we go down that road, then I know people who claim being anywhere within blocks of a cellphone tower or in a building with wifi has ruined their lives. There is a movement (I think in the UK) to remove and disallow wifi on all universuty premises because a few people have said they are electrosensitive. If this is anything like peanut allergies, it's no joke for those afflicted, but before I rip out our DECT phone, I'll need to find a compelling reason. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Re: OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 9:51 PM, Steve Totaro [EMAIL PROTECTED] wrote: Digium is so bigtime they should be giving away re-branded (acid etched so it doesn't rub off) leatherman tools to everyone who has ever ordered from them ;-) You get those for every order of 10 or more ABE! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
I use them for analog Polycom Soundstation EXes and other analog conference phones. Great quality and that is in boardrooms where complaints would fly if there was even one little issue. I generally don't mess with Grandstream but the ATAs aren't bad. Thanks, Steve T On Tue, Jun 17, 2008 at 3:59 PM, Fred Posner [EMAIL PROTECTED] wrote: They're real cheap where you could get 2 of them. I got one and actually have no complaints. Call quality was really good and it's very, very small and portable. Looks cheap, but you get over that. I have a SPA2102 2 liner which works fine but gets really hot. Fred Posner www.teamforrest.com FWD#: 902963 On Jun 17, 2008, at 3:46 PM, Eric Fort wrote: and being only a single line device how exactly would I get 2 lines out of it? Eric On Tue, Jun 17, 2008 at 12:22 PM, Steve Totaro [EMAIL PROTECTED] wrote: I like Grandstream 286s No, seriously Thanks, Steve T On Tue, Jun 17, 2008 at 2:07 PM, Eric Fort [EMAIL PROTECTED] wrote: Can the PAP2 be set up such that a second call will ring the second line when the first is busy but only register once with the SIP provider? A beep tone on the same line to denote another incoming call just will not do, The second port needs to act like a seperate line tied to the same DID in a hunt group. Eric On Tue, Jun 17, 2008 at 3:52 AM, Rob Hillis [EMAIL PROTECTED] wrote: IMO, yes - sort of. :) Since Linksys bought Sipura, you're probably looking at the Linksys PAP2 - the functional equivalent of the Sipura SPA-2000. They look different (better if you ask me - the LEDs are far better placed and more useful than they were on the Sipura units) but are pretty much identical under the hood. randulo wrote: On Tue, Jun 17, 2008 at 7:37 AM, Gordon Henderson [EMAIL PROTECTED] wrote: But maybe an AVM Fritz! box will work for you too... Would anyone care to recommend a good quality, stable ATA these days for just a single cordless phone connected to one SIP provider. Sipura used to be well thought-of. Are they still the best? /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:485753dc40256671610936! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] need ata suggestion
On Tue, Jun 17, 2008 at 4:06 PM, randulo [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 9:56 PM, Steve Totaro [EMAIL PROTECTED] wrote: DECT can make you and others around you feel very ill. No long term research exists but if immediate effects are feeling ill, it may possibly lead to long term effects. That may or may not be true, but if we go down that road, then I know people who claim being anywhere within blocks of a cellphone tower or in a building with wifi has ruined their lives. There is a movement (I think in the UK) to remove and disallow wifi on all universuty premises because a few people have said they are electrosensitive. If this is anything like peanut allergies, it's no joke for those afflicted, but before I rip out our DECT phone, I'll need to find a compelling reason. A compelling reason to me would be if someone near me felt ill, I switched off the DECT (quietly) and then they felt better, then switched it back on and see if they complain again, if not I would ask How are you feeling? If ill, that would be compelling enough to me. Maybe repeat a few times for consistency. No complaints, then no problem Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] suggestions for IAX ATA device or phone in US
anyone has used or bough one? would appreciate comments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Hello, If you have a PRI-T1 in the USA, then you can set outgoing CallerID with just about any carrier. MATT--- On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote: How can they even set such 1234567890 callerIDs anyway? For example, our inter/intra state calling depends a lot on the callerIDs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: June 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
I can set to anything on my Qwest circuit. All zeros or whatever, just has to be ten digits. I have seen some that will send less than ten like a four digit extension number on a misconfigured system. Thanks, Steve T On Tue, Jun 17, 2008 at 4:38 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, If you have a PRI-T1 in the USA, then you can set outgoing CallerID with just about any carrier. MATT--- On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote: How can they even set such 1234567890 callerIDs anyway? For example, our inter/intra state calling depends a lot on the callerIDs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: June 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
My questions was to the fact that JRA mentioned he knows at least 3 owners. to which I asked if it was LLCs or other type of corporations, since LLCs have different rules. What I mentioned about it being illegal is for non LLC type of corporations, but for most of the other types of corporations, while it's possible that it is illegal for LLCs as well in some states I could understand that the rules could be relaxed for LLCs as well. As far as the IRS goes i'm quite positive that it's illegal for tax purposes, in other words it cannot be counted as a business expense. The way I understand this: http://www.irs.gov/businesses/small/article/0,,id=146835,00.html towards the bottom of the page, it cannot always be used as a business expense. Correct me if I'm wrong. On Tue, Jun 17, 2008 at 12:21 PM, Steve Totaro [EMAIL PROTECTED] wrote: You are probably confusing corporate tactics to pay less taxes vs corporate tactics to protect assets. The first does provide some asset protection but is mainly to pay less taxes. The second is to basically hide assets through totally legal LLCs. Thanks, Steve Totaro On Tue, Jun 17, 2008 at 12:00 PM, C F [EMAIL PROTECTED] wrote: LLCs? On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zapata/DAHDI Disable Hardware echo canceler based on SDA number / displan
On Tue, Jun 17, 2008 at 10:56:06AM -0500, Kevin P. Fleming wrote: Benoit Plessis wrote: Is it possible on a TE220p to deactivate the hardware echo canceler at will ? (With a function in the dialpan for example) example for fax SDA ,beeing able to shutdown the echo canceler could give better results don't you think ? All echo cancelers using Zaptel/DAHDI already disable themselves when FAX or modem communications are used, based on reception and detection of the CED tone that FAX machines and modems generate to make that happen. You can tell this happened by looking at the channel in Asterisk using 'zap show channel' or 'dahdi show channel' as it will show you that the echo canceler was disabled automatically. Karamba ! It mean than that my last option is to put all three digium cards in one box and hope, that it'll fit and that it'll work better :( (actually i'm using one server with two cards (TE220, B410) which is linked with an (slinear) IAX peer to another server with one TDM800 to talk to the fax machine. -- Benoit ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Yeah, but what do you get billed as? I understand if your callerID and the called party is from within a state, it's interstate routing. If between states, then it's intrastate, etc The billing depends on the callerID you send. So, if you send a 000-000- clid to a 917 area code, what would the call be routed/billed as? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 17, 2008 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! I can set to anything on my Qwest circuit. All zeros or whatever, just has to be ten digits. I have seen some that will send less than ten like a four digit extension number on a misconfigured system. Thanks, Steve T On Tue, Jun 17, 2008 at 4:38 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, If you have a PRI-T1 in the USA, then you can set outgoing CallerID with just about any carrier. MATT--- On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote: How can they even set such 1234567890 callerIDs anyway? For example, our inter/intra state calling depends a lot on the callerIDs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: June 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's jurisdiction. Thanks, Steve T On Fri, Jun 13, 2008 at 6:15 AM, Matt Florell [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
If it shows up as the BTN on the CDRs then technically you should be billed at the highest possible tariff. Whether your provider will do that or not depends what they are charged. In general the provider/s shouldn't use CID as the BTN and therefore you shouldn't be over or under charged. Even in cases where the CID is actually passed along as the BTN, the provider should still keep track of you by circuit ID rather than CID, however when they have to pay their tariffs I am assuming they will be charged based on BTN which they based on CID that you set, which will in turn make them lose money IF they are charged at highest possible tariff. In conclusion, I don't know what you are charged because I haven't seen your bills. I don't know if the providers actually have the capabilities to do BTN different than CID (I am assuming they could), and if they do have the capablity they should actually make sure that the BTN is always set to what it is and not CID. If they don't they should pass on any high tariffs resulting from that to you. On Tue, Jun 17, 2008 at 5:30 PM, Mark Hamilton [EMAIL PROTECTED] wrote: Yeah, but what do you get billed as? I understand if your callerID and the called party is from within a state, it's interstate routing. If between states, then it's intrastate, etc The billing depends on the callerID you send. So, if you send a 000-000- clid to a 917 area code, what would the call be routed/billed as? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 17, 2008 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! I can set to anything on my Qwest circuit. All zeros or whatever, just has to be ten digits. I have seen some that will send less than ten like a four digit extension number on a misconfigured system. Thanks, Steve T On Tue, Jun 17, 2008 at 4:38 PM, Matt Florell [EMAIL PROTECTED] wrote: Hello, If you have a PRI-T1 in the USA, then you can set outgoing CallerID with just about any carrier. MATT--- On 6/17/08, Mark Hamilton [EMAIL PROTECTED] wrote: How can they even set such 1234567890 callerIDs anyway? For example, our inter/intra state calling depends a lot on the callerIDs. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: June 13, 2008 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! Hello, I am not suggesting that the USA's laws exist outside of the USA, I can imagine the horrible problems that would cause in the rest of world. I wanted to point out that if you are using this service and doing business in the USA that you could face penalties for not following the law. According to the FTC, both companies(the scrubber and the client) are guilty of breaking the laws of the USA. If you are calling the USA and need to use this company's FTC DNC list filtering services then you may have USA-based operations of some kind. In such cases it is important to note that companies have been fined millions of dollars and have been shut down in the USA for violating these regulations. I am well aware of the fact that companies based outside of the USA routinely call-blast the USA with auto-dialers that send out callerIDs such as 1234567890 and do no filtering against the USA FTC DNC lists. A large portion of these companies are doing lead-generation for USA-based companies, and over the years a lot of those USA-based companies have been shut down for the activities of their lead suppliers. MATT--- On 6/13/08, Dean Collins [EMAIL PROTECTED] wrote: Yep it's funny how few people on this list realize that the usa's borders and laws stop 50 miles off the coast. It's also surprising how few Americans realize that a company incorporated internationally (Pakistan in this instance) even if owned as a subsidiary of a USA parent doesn't have to follow the laws of the USA but actually falls under the jurisdiction of the laws they are incorporated under. I'm not saying this is good or bad, 'm just saying that as 'asterisk' people we should be smart enough to play the laws that suit us to our advantage, if you think that the Global 1000 companies don't then you are kidding yourself. Besides we have the advantage in that almost everything we do can be virtual in most instances. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, 13 June 2008 7:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My guess is that they are outside of the FTC's
[asterisk-users] GXW 4108 asterisk configuration
Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn't work. Someone has a good configuration for this gateway? Thanks in advance, Nelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
Hi, I'm installing zaptel-source_1.4.10.1~dfsg-1_all.deb (from Debian SID) into my ubuntu 8.04 box with: dpkg -i zaptel-source_1.4.10.1~dfsg-1_all.deb ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel Loading the wcfxo module and/or zaptel: [EMAIL PROTECTED]:~# modprobe wcfxo WARNING: Error inserting zaptel (/lib/modules/2.6.24-16-server/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting wcfxo (/lib/modules/2.6.24-16-server/misc/wcfxo.ko): Unknown symbol in module, or unknown parameter (see dmesg) [EMAIL PROTECTED]:~# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.24-16-server/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) When the build is finished and the box restarted, I'm getting this dmesg output: [ 46.160498] zaptel: Unknown symbol oslec_echo_can_identify [ 46.180909] ztdummy: Unknown symbol zt_receive [ 46.181054] ztdummy: Unknown symbol zt_transmit [ 46.181126] ztdummy: Unknown symbol zt_unregister [ 46.181221] ztdummy: Unknown symbol zt_register [ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify [ 830.122738] wcfxo: Unknown symbol zt_receive [ 830.122890] wcfxo: Unknown symbol zt_ec_chunk [ 830.123037] wcfxo: Unknown symbol zt_transmit [ 830.123112] wcfxo: Unknown symbol zt_unregister [ 830.123212] wcfxo: Unknown symbol zt_hooksig [ 830.123301] wcfxo: Unknown symbol zt_register [ 830.123377] wcfxo: Unknown symbol zt_alarm_notify [ 858.887084] zaptel: Unknown symbol oslec_echo_can_identify [EMAIL PROTECTED]:~# This is the modinfo output: [EMAIL PROTECTED]:~# modinfo zaptel filename: /lib/modules/2.6.24-16-server/misc/zaptel.ko version:1.4.10.1 license:GPL description:Zapata Telephony Interface author: Mark Spencer [EMAIL PROTECTED] srcversion: 927BA7DCB504C0BA7C0CDED depends:oslec,crc-ccitt vermagic: 2.6.24-16-server SMP mod_unload 686 parm: debug:int parm: deftaps:int [EMAIL PROTECTED]:~# modinfo wcfxo filename: /lib/modules/2.6.24-16-server/misc/wcfxo.ko license:GPL author: Mark Spencer [EMAIL PROTECTED] description:Wildcard X100P Zaptel Driver srcversion: 194D48A51D46F480234E26A alias: pci:v1057d5608sv*sd*bc*sc*i* alias: pci:vE159d0001sv8087sd*bc*sc*i* alias: pci:vE159d0001sv8086sd*bc*sc*i* alias: pci:vE159d0001sv8085sd*bc*sc*i* alias: pci:vE159d0001sv8084sd*bc*sc*i* depends:zaptel vermagic: 2.6.24-16-server SMP mod_unload 686 parm: debug:int parm: quiet:int parm: boost:int parm: monitor:int parm: opermode:int [EMAIL PROTECTED]:~# modinfo oslec filename: /lib/modules/2.6.24-16-server/oslec.ko description:Open Source Line Echo Canceller Zaptel Wrapper author: David Rowe license:GPL srcversion: 9C9E87427F162644A61A1CB depends: vermagic: 2.6.24-16-server SMP mod_unload 686 I've the same trouble installing from sources and patching the zaptel sources with oslec. I've installed before on debian sarge/etch and ubuntu 7.10 without problems. What can be wrong? Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
Guillermo Salas M. wrote: [ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify Make sure you get the latest version of OSLEC from SVN - the downloadable tarball has a bug in it which prevents it from compiling properly (although it acts like it worked just fine); which then prevents zaptel from loading. If it all still fails, try going back to a slightly earlier version of Zaptel (1.4.9.2). Basically, follow the instructions here: http://www.rowetel.com/ucasterisk/oslec.html (the HowTo - Run OSLEC with Asterisk/Zaptel section) HTH! Cheers, Ade. No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.3.0/1505 - Release Date: 16/06/2008 07:20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
Hi, See comments in-line On Tue, Jun 17, 2008 at 04:56:53PM -0500, Guillermo Salas M. wrote: Hi, I'm installing zaptel-source_1.4.10.1~dfsg-1_all.deb (from Debian SID) into my ubuntu 8.04 box with: dpkg -i zaptel-source_1.4.10.1~dfsg-1_all.deb ECHO_CAN_NAME=OSLEC m-a -t a-i zaptel Actually if you look at the changelog entry for that version you'll see: '* Set OSLEC as the default echo canceller.' The default build is now with OSLEC. Loading the wcfxo module and/or zaptel: [EMAIL PROTECTED]:~# modprobe wcfxo WARNING: Error inserting zaptel (/lib/modules/2.6.24-16-server/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting wcfxo (/lib/modules/2.6.24-16-server/misc/wcfxo.ko): Unknown symbol in module, or unknown parameter (see dmesg) [EMAIL PROTECTED]:~# modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.24-16-server/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) When the build is finished and the box restarted, I'm getting this dmesg output: [ 46.160498] zaptel: Unknown symbol oslec_echo_can_identify [ 46.180909] ztdummy: Unknown symbol zt_receive [ 46.181054] ztdummy: Unknown symbol zt_transmit [ 46.181126] ztdummy: Unknown symbol zt_unregister [ 46.181221] ztdummy: Unknown symbol zt_register That's odd. zaptel should depend on oslec and pull it on modprobe. [ 830.118287] zaptel: Unknown symbol oslec_echo_can_identify [ 830.122738] wcfxo: Unknown symbol zt_receive [ 830.122890] wcfxo: Unknown symbol zt_ec_chunk [ 830.123037] wcfxo: Unknown symbol zt_transmit [ 830.123112] wcfxo: Unknown symbol zt_unregister [ 830.123212] wcfxo: Unknown symbol zt_hooksig [ 830.123301] wcfxo: Unknown symbol zt_register [ 830.123377] wcfxo: Unknown symbol zt_alarm_notify [ 858.887084] zaptel: Unknown symbol oslec_echo_can_identify [EMAIL PROTECTED]:~# This is the modinfo output: [EMAIL PROTECTED]:~# modinfo zaptel filename: /lib/modules/2.6.24-16-server/misc/zaptel.ko version:1.4.10.1 license:GPL description:Zapata Telephony Interface author: Mark Spencer [EMAIL PROTECTED] srcversion: 927BA7DCB504C0BA7C0CDED depends:oslec,crc-ccitt zaptel does depend on oslec... So what can it be? vermagic: 2.6.24-16-server SMP mod_unload 686 parm: debug:int parm: deftaps:int [EMAIL PROTECTED]:~# modinfo wcfxo filename: /lib/modules/2.6.24-16-server/misc/wcfxo.ko license:GPL author: Mark Spencer [EMAIL PROTECTED] description:Wildcard X100P Zaptel Driver srcversion: 194D48A51D46F480234E26A alias: pci:v1057d5608sv*sd*bc*sc*i* alias: pci:vE159d0001sv8087sd*bc*sc*i* alias: pci:vE159d0001sv8086sd*bc*sc*i* alias: pci:vE159d0001sv8085sd*bc*sc*i* alias: pci:vE159d0001sv8084sd*bc*sc*i* depends:zaptel vermagic: 2.6.24-16-server SMP mod_unload 686 parm: debug:int parm: quiet:int parm: boost:int parm: monitor:int parm: opermode:int [EMAIL PROTECTED]:~# modinfo oslec filename: /lib/modules/2.6.24-16-server/oslec.ko description:Open Source Line Echo Canceller Zaptel Wrapper author: David Rowe license:GPL srcversion: 9C9E87427F162644A61A1CB depends: vermagic: 2.6.24-16-server SMP mod_unload 686 That's a strange place. Is there /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko ? find /lib/modules/2.6.24-16-server/ -name oslec.ko I suspect there's an older and incompatible copy of oslec.ko around. I've the same trouble installing from sources and patching the zaptel sources with oslec. I've installed before on debian sarge/etch and ubuntu 7.10 without problems. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center
Syed Nasruddin wrote: Dear Sherwood, Thanks. Just three questions: 1. Will I be needing Apache or Asterk-stat will handle itself? 2. Are there How-tos for integerating asterisk-stat with asterisk? 3. My Recordings are being saved in the default folder i.e: /var/spool/asterisk/monitor/ in .gsm format. When I wish to listen to a particular recording I first convert it with SOX utility into .wav format and then listen it. Will this also be automated so that when I select a recording and try to listen it will be in right format. Thanks again. Syed Nasruddin Apache/PHP and the appropriate database plugin will be needed :) Asterisk-Stat comes with installation information Yes, but only if you modify the Asterisk-Stat's pages as I have done, by creating a listen/download link in the system and having it link to the recording in question. You can also make the link call a shell script that transcodes the file before offering it up. -- Sherwood McGowan VoIP / Telecom Solutions [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone have pricing on the Color Polycom Phone?
IP670 was just released...about 30% more than the IP650. http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/soundpoint_ip670.html -Matt On Tue, Apr 29, 2008 at 1:02 AM, Patrick [EMAIL PROTECTED] wrote: On Mon, 2008-04-28 at 14:49 -1000, Matt Darnell wrote: Anyone seen anything on the IP670 the Color Expansion? Great timing. Yesterday I was looking at the IP650 and wondered when the successor to the IP650 would arrive. Do you have a link or more info about the IP670? Thanks, Patrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
El mié, 18-06-2008 a las 01:37 +0300, Tzafrir Cohen escribió: That's a strange place. Is there /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko ? find /lib/modules/2.6.24-16-server/ -name oslec.ko I suspect there's an older and incompatible copy of oslec.ko around. You are right: find /lib/modules/2.6.24-16-server/ -name oslec.ko /lib/modules/2.6.24-16-server/oslec.ko /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko I will be deleting all oslec.ko references, modules/zaptel directory and start again. Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.10.1 and OSLEC on Ubuntu 8.04
El mar, 17-06-2008 a las 18:01 -0500, Guillermo Salas M. escribió: find /lib/modules/2.6.24-16-server/ -name oslec.ko /lib/modules/2.6.24-16-server/oslec.ko /lib/modules/2.6.24-16-server/misc/oslec/oslec.ko I will be deleting all oslec.ko references, modules/zaptel directory and start again. It works :) [EMAIL PROTECTED]:~# lsmod | grep oslec oslec 10396 1 zaptel [EMAIL PROTECTED]:~# lsmod | grep zaptel zaptel195588 6 wcfxo,wcopenpci oslec 10396 1 zaptel crc_ccitt 3072 2 zaptel,hisax Thank you very much for your help. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help! - Double NAT issue
Try this. It WFM: localnet=10.0.0.0/255.255.255.0 nat = yes stunaddr = stun.ekiga.net ; or some other stun server, e.g.: foo.stun.com:3478 externrefresh = 15 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] adding funcionatlity to asterisk?! is it possible?!
Right now the issue I see is you are using overlapping extensions
so maybe that's not working as expected?
you have in context sipura line exten 201, exten 201 included from
context spa and also exten 2xx included from context spa.
What you want to do with sending calls elsewhere if they are not
completed look at DIALSTATUS, e,g,:
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} - SIP DEVICE
; ${ARG2} - ringing seconds
; ${ARG3} - vm-box-Nr.
;
exten = s,1,Macro(docid)
exten = s,2,Dial(SIP/${ARG1},${ARG2},r); Ring
the ${ARG1} interface, ${ARG3} seconds maximum
exten = s,3,Goto(s-${DIALSTATUS},1); Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Playback(silence/1)
exten = s-NOANSWER,2,Voicemail(${ARG3},u) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,3,Goto(s-${VMSTATUS},1)
exten = s-USEREXIT,1,Playback(cancelled)
exten = s-USEREXIT,2,Playback(goodbye)
exten = s-USEREXIT,3,Hangup
exten = s-SUCCESS,1,Playback(goodbye)
exten = s-SUCCESS,2,Hangup
exten = s-FAILED,1,Playback(sorry-youre-having-problems)
exten = s-FAILED,2,Playback(please-try-again-later)
exten = s-FAILED,3,Playback(goodbye)
exten = s-FAILED,4,Hangup
exten = o-CHANUNAVAIL,1,Goto(o-BUSY,1)
exten = s-BUSY,1,Playback(silence/1)
exten = s-BUSY,2,Voicemail(${ARG3},b) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,3,Playback(goodbye) ; If they press #,
return to start
exten = s-BUSY,4,Hangup
exten = o,1,Goto(o-${DIALSTATUS},1)
exten = _o-.,1,Goto(o-NOANSWER,1)
exten = o-BUSY,1,Goto(s,2)
exten = o-NOANSWER,1,Playback(please-try-again)
exten = o-NOANSWER,2,GoTo(s-NOANSWER,2)
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else
as no answer
exten = a,1,Playback(this-is-the-voice-mail-system)
exten = a,2,VoicemailMain(${ARG3}) ; If they press *,
send the user into VoicemailMain
For the directory, there's a directory application built into the
voicemail system. You might want to check that out, if it fits your
needs then it's probably the simplest solution.
On Sat, Jun 14, 2008 at 5:56 AM, RoLaNd RoLaNd [EMAIL PROTECTED] wrote:
hello all,
im looking for a way to do the following:
when a SPECIFIC call comes through to asterisk through sip, i want it to b
directed to a pool of specific sip extensions (9 extensions) where asterisk
tries one after the other till lhe finds one of them thats actually on.
i want to add a step for asterisk to follow which is, when a sip extension
doesn't answer or its offline, instead of immediately transferring to voice
mail, i want it to dial that sip holder's number so it transfers the call to
his cellphone for example. and if he didn't answer his cellphone its then
that i want it to direct it to voice mail.
i want to add another item to the operator menu, instead of just receiving
the call and telling the caller to either dial extension or 100 for
operator, i want asterisk to offer the caller an additional option like for
example pressing 2, would direct you to a list of key personnels with their
respective extensions.
please find below my extensions.conf:
[sipura-line]
exten = 201,1,Answer() ; Answer inbound calls
exten = 201,2,Playback(silence/1)
exten = 201,3,Background(simzy1) ; input an extension
exten = 201,4,Wait(8)
include = spa
exten = 201,n,Hangup()
[spa]
exten =_201,1,GoTo(sipura-line,${EXTEN},1)
exten = _1XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED])
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},15) ;each ring equals to 5 seconds, so it
will ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED])
exten = _2XX,3,HangUp()
exten =_01,1,Dial(SIP/200)
exten = 203,1,VoicemailMain
exten = _2XX,1,Dial(SIP/${EXTEN},15)
Invite your mail contacts to join your friends list with Windows Live
Spaces. It's easy! Try it!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange SIP-SIP delay
On Tue, Jun 17, 2008 at 11:39 AM, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: I've got the following setup: PhoneA - router - vpn - router- asterisk (SIP / ISDN) PhoneB - asterisk (SIP / ISDN) If phone A is connected to phone B (sip-sip), there is a noticable delay (up to 2-3 seconds) between me speaking and the other end hearing. If phone A calls out via the ISDN and back in to the DDI of phone B (ie SIP-ISDN-ISDN-SIP) then there is no delay at all ! Any clues on where I might start looking for this ? Are you using canreinvite=yes setting (i.e. is the RTP media expected to flow directly between the phones as opposed to hair-pining through Asterisk)? If so, some of the delay could be attributed to the time spent in RE-INVITEs that happen after the call set up. -- Raj Jain P.S. There is the directrtpsetup= flag that can eliminate this latency (if you're indeed using canreinvite=yes), but I believe that feature is considered experimental. Actually, if that feature is still experimental, I'd like to change that and fix any associated bugs because it seems like a pretty useful feature to me for people who want to use Asterisk as a call controller (a.k.a. soft-switch) that does not need to participate in the media path. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
Bzzzt I'm not an accountant, and don't play one on tv but you are wrong. This only relates to the classification of the income as passive and has nothing to do with can a director of a business shield himself. Go pay someone $250 an hour and they'll tell you how it affects you and stop wasting electrons on this sill email chain. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, 17 June 2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My questions was to the fact that JRA mentioned he knows at least 3 owners. to which I asked if it was LLCs or other type of corporations, since LLCs have different rules. What I mentioned about it being illegal is for non LLC type of corporations, but for most of the other types of corporations, while it's possible that it is illegal for LLCs as well in some states I could understand that the rules could be relaxed for LLCs as well. As far as the IRS goes i'm quite positive that it's illegal for tax purposes, in other words it cannot be counted as a business expense. The way I understand this: http://www.irs.gov/businesses/small/article/0,,id=146835,00.html towards the bottom of the page, it cannot always be used as a business expense. Correct me if I'm wrong. On Tue, Jun 17, 2008 at 12:21 PM, Steve Totaro [EMAIL PROTECTED] wrote: You are probably confusing corporate tactics to pay less taxes vs corporate tactics to protect assets. The first does provide some asset protection but is mainly to pay less taxes. The second is to basically hide assets through totally legal LLCs. Thanks, Steve Totaro On Tue, Jun 17, 2008 at 12:00 PM, C F [EMAIL PROTECTED] wrote: LLCs? On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Invitation to connect on LinkedIn
LinkedIn Vinicius Bossle Fagundes requested to add you as a connection on LinkedIn: -- Ricardo, I'd like to add you to my professional network on LinkedIn. -Vinicius View invitation from Vinicius Bossle Fagundes http://www.linkedin.com/e/IUZTDdzrsg3rxGytdedLzTiomUEFOT3UdcnGbWCo8rrTM7G/blk/620032048_2/cBYUd30OcP0MczoLqnpPbOYWrSlI/svi/ -- DID YOU KNOW you can showcase your professional knowledge on LinkedIn to receive job/consulting offers and enhance your professional reputation? Posting replies to questions on LinkedIn Answers puts you in front of the world's professional community. http://www.linkedin.com/e/abq/inv-24/ -- (c) 2008, LinkedIn Corporation ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Canadian Whitepage Listing Capability
So my SIP Provider states they do not offer the service to list my numbers w/ the Whitepages. We phoned the Whitepages and they said we can't do it, the SIP Provider must? Either one/both of them is/are useless or I must switch SIP providers to one that can get this done. Anyone familiar with this fiasco and can help steer me in the right direction? Any suggestions would be greatly appreciated! Thanks, jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] World Cheapest Predictive Dialer!
On Tue, Jun 17, 2008 at 7:57 PM, Dean Collins [EMAIL PROTECTED] wrote: Bzzzt I'm not an accountant, and don't play one on tv but you are wrong. This only relates to the classification of the income as passive and has nothing to do with can a director of a business shield himself. Go pay someone $250 an hour and they'll tell you how it affects you and stop wasting electrons on this sill email chain. Like I said in the last email, I misread it and it has nothing to do with if a director of a business may rent his property to his business. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, 17 June 2008 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] World Cheapest Predictive Dialer! My questions was to the fact that JRA mentioned he knows at least 3 owners. to which I asked if it was LLCs or other type of corporations, since LLCs have different rules. What I mentioned about it being illegal is for non LLC type of corporations, but for most of the other types of corporations, while it's possible that it is illegal for LLCs as well in some states I could understand that the rules could be relaxed for LLCs as well. As far as the IRS goes i'm quite positive that it's illegal for tax purposes, in other words it cannot be counted as a business expense. The way I understand this: http://www.irs.gov/businesses/small/article/0,,id=146835,00.html towards the bottom of the page, it cannot always be used as a business expense. Correct me if I'm wrong. On Tue, Jun 17, 2008 at 12:21 PM, Steve Totaro [EMAIL PROTECTED] wrote: You are probably confusing corporate tactics to pay less taxes vs corporate tactics to protect assets. The first does provide some asset protection but is mainly to pay less taxes. The second is to basically hide assets through totally legal LLCs. Thanks, Steve Totaro On Tue, Jun 17, 2008 at 12:00 PM, C F [EMAIL PROTECTED] wrote: LLCs? On 6/16/08, Jay R. Ashworth [EMAIL PROTECTED] wrote: On Sat, Jun 14, 2008 at 11:13:31PM -0400, C F wrote: Happens in the commercial world all the time; it's a common way to get cash out of the corporation -- a business's building is owned by the corporation's owners, and rented to the corporation. This is actually illegal in some states and considered a breach of Fiduciary everywhere. May be, but I know at least 3 owners of private corporations who are doing it, and their auditors seem fine with it. I think that it matters whether the corporation is public or not... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canadian Whitepage Listing Capability
Joseph L. Casale wrote: So my SIP Provider states they do not offer the service to list my numbers w/ the Whitepages. We phoned the Whitepages and they said we can't do it, the SIP Provider must? Either one/both of them is/are useless or I must switch SIP providers to one that can get this done. Anyone familiar with this fiasco and can help steer me in the right direction? Any suggestions would be greatly appreciated! I am not aware of any ITSPs (Internet phone companies) that provide white pages listings. They could exist, but I doubt it. -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Canadian Whitepage Listing Capability
Joseph L. Casale wrote: So my SIP Provider states they do not offer the service to list my numbers w/ the Whitepages. We phoned the Whitepages and they said we can't do it, the SIP Provider must? Either one/both of them is/are useless or I must switch SIP providers to one that can get this done. Anyone familiar with this fiasco and can help steer me in the right direction? Any suggestions would be greatly appreciated! In BC I've been able to get listings added by calling Superpages and asking them for an Additional listing. Last time I checked they billed something like $3.35/month *plus* an insertion fee. Even though I dealt with Superpages directly to generate the listing, my bills came from Telus (I don't have any Telus services). I'm not sure what province you're in, but maybe those clues will help point you in the right direction. Trevor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
