[asterisk-users] unistim channel problem

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0

[asterisk-users] Stress Testing IVR

Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be "programmed" to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendatio

Re: [asterisk-users] Credit Card processing machines

On Fri, Feb 6, 2009 at 17:11, Jeff LaCoursiere wrote: > > Anyone have much luck with these on ATA's? I have a few sites that use > them succesfully with multi-port Audiocodes boxes, but just connected ten > machines to Linksys 2102s and they are very flaky. Using u-law on a 100Mb > switched netw

[asterisk-users] TeleKaam - Voice Portal for Students and Parents

Hello All We have started a voice portal for Parents and Students. They can listen Grades, Attendance status and other relevant information over *phone*. Please read features below and to listen Demo IVR call at *00 92 42 8315427. *Initially we are deploying it for a School and planning to spread

Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

Looks like my provider is not passing dtmf correctly .. Had a serious laugh, system kept asking me if I was ready., ended up finding myself talking to the IVR . On Mon, Feb 16, 2009 at 11:45 AM, Asterisk Asterisk wrote: > This module detects gender and approximate age range. I'm wor

[asterisk-users] Optimizing this script for calling Skype users from Asterisk

I have written this configuration script which uses OpenSky to make Skype calls directly from Asterisk devices using my companies SIP to Skype gateway. Users can dial skype_anyskypeusername or manually add names or extensions which can get mapped to the correct dialing sequence. The right sequence

Re: [asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)

The old classic is to say something like ' your callerid is blocked, please get out your credit card' PaulH Alfred Monticello wrote: > I'm thinking of starting a partyline, where people call in and talk to > other people. For record keeping and billing purposes, I'd like to go > by the callers

[asterisk-users] Obtaining callerid on a PRI for billing purposes (with non toll-free numbers)

I'm thinking of starting a partyline, where people call in and talk to other people. For record keeping and billing purposes, I'd like to go by the callers telephone number. This method works fine as long as the caller doesn't have callerid blocked, but what are my options if they do block the

[asterisk-users] mp3player() to shuffle playlist

Hi all, I am trying to make a scenario when someone dial *10*, the mp3player() function would act and play a list of MP3 files. However, I have no idea how to randomize the function (mpg123 is capable of shuffling the MP3 files, buat how to implement it in extensions.conf?) Perhaps any of you su

Re: [asterisk-users] DTMF not completely muted

Wilton Helm compuserve.com> writes: > The DADHI function is probably intended for more > generalized use. Maybe for recording voicemail greetings it should not be > used and a different function used instead. There is no reason why it > isn't possible to backup in the recorded message and er

Re: [asterisk-users] OpenSky: Digium Skype gateway?

What about receiving Skype calls on Gizmo or other SIP device? Looking into the website I don't see anything regarding that. On Mon, Feb 16, 2009 at 8:05 AM, Olivier wrote: > > > 2009/2/13 John Todd >> >> On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote: >> >> > Hi there, >> > >> > is g

Re: [asterisk-users] DTMF not completely muted

The DADHI function is probably intended for more generalized use. Maybe for recording voicemail greetings it should not be used and a different function used instead. There is no reason why it isn't possible to backup in the recorded message and erase the blip. The detection time should appro

Re: [asterisk-users] WiFi SIP phone w/VPN?

Ken D'Ambrosio wrote: > Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN > abilities? Failing that, a WiFi phone that runs Linux? I already know > one phone that does meet my requirements -- the iPhone. The new software > comes with a Cisco VPN client, and a SIP client c

[asterisk-users] How to beep before transfer ...

Hi, When I transfer a call to an extension, the person I call does not have any idea when that transfer happened so it is a guessing game. Is there a way to send a beep to the caller just before transferring the call? Preferably by setting something in FreePbx?   Sincerely, robert _

Re: [asterisk-users] [OT] Gmail is broken

On 16/02/2009 10:23 p.m., Grygoriy Dobrovolskyy wrote: > Sorry about off-topic, but can you advise the mail client who is able to > organise the web mailing list topic as web interface does ? (i mean by > blocks/topics) I wold be glad to use something else with the same usability, > but dont see an

Re: [asterisk-users] Michael Graves post

On 17/02/2009 3:05 a.m., Michael Graves wrote: > Phasing tricks using stereo speakers are good as far as generating an > effect. That is, synthesizing the perception of some image width, but > not accurately reproducing an audible scene. There are many fine > commercial examples of this. I own a ol

Re: [asterisk-users] DTMF not completely muted

Kevin P. Fleming digium.com> writes: > Michael Smith wrote: > > > When the Dahdi driver detects DTMF, it seems it's not muting the first > > 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. > > The blip at the beginning is expected; the DTMF detector won't trigger > until it has seen a

Re: [asterisk-users] Please help test the gender detection module at 575-613-4392

This module detects gender and approximate age range. I'm working on getting it's accuracy to 80%+ on a consistent basis, after implementing filters to remove background noise and other artifacts. It's designed for a number of things. To start, I have several clients (primarily mobile content a

Re: [asterisk-users] DTMF not completely muted

Michael Smith wrote: > When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms > and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded > voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, > at the end of the recording. The bl

[asterisk-users] vmsecret question

Hi! I wonder what is teh meaning if vmsecret is not defined. Does this mean that the voicebox can be accessed without PIN code? Or does it mean that the voicebox can not be accessed at all (of course except using the s Parameter of VoicemailMain()) ? thanks klaus _

[asterisk-users] DTMF not completely muted

Hi all, When the Dahdi driver detects DTMF, it seems it's not muting the first 5-15 ms and sometimes the last 2-10 ms of the DTMF tone. This shows up in recorded voicemail greetings -- you hear a very short DTMF '#', or sometimes two blips, at the end of the recording. I have a Mitel SX-200 conne

[asterisk-users] Please help test the gender detection module at 575-613-4392

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to thr

Re: [asterisk-users] command show channels concise

Jerry Geis wrote: > I am getting a priveldged command error on the manager API. > > 16-Feb-09 11:51 am asterisk_command() Action: Login > 16-Feb-09 11:51 am asterisk_command() Username: XXX > 16-Feb-09 11:51 am asterisk_command() Secret: > 16-Feb-09 11:51 am asterisk_command() Events: off > 1

Re: [asterisk-users] AstDB wildard searches

On Monday, February 16, 2009, Jared Smith wrote: > Hopefully that helps make things a bit more clear. It does - many thanks for your help. -- Geoff ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing lis

Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

On Feb 13, 2009, at 9:59 AM, John Todd wrote: > > I've been involved with getting better data for running Asterisk on > the Amazon EC2 cloud computing system. Here are some calculations > I've made on costs based on current published prices on Amazon's > system. Feel free to tell me that I'm wr

Re: [asterisk-users] AstDB wildard searches

On Mon, 2009-02-16 at 16:30 +, Geoff Lane wrote: > Thanks for your suggestion. My Asterisk machine has MySQL, and I might > go down that route at some time. However, my query is part of an > almost academic exercise in which I'm trying to find out what AstDB is > capable of. Unfortunately your

[asterisk-users] command show channels concise

I am getting a priveldged command error on the manager API. 16-Feb-09 11:51 am asterisk_command() Action: Login 16-Feb-09 11:51 am asterisk_command() Username: XXX 16-Feb-09 11:51 am asterisk_command() Secret: 16-Feb-09 11:51 am asterisk_command() Events: off 16-Feb-09 11:51 am DEBUG: Respons

Re: [asterisk-users] AstDB wildard searches

On Monday, February 16, 2009, Jared Smith wrote: > If you have that many items in a database and want to do those types > of filters, why not stick them in a SQL database and use func_odbc > to retrieve them from your SQL database inside the dialplan? Thanks for your suggestion. My Asterisk machi

Re: [asterisk-users] AstDB wildard searches

On Mon, 2009-02-16 at 15:14 +, Geoff Lane wrote: > I'm looking for a way to filter the AstDB cidname family to show only > those entries with a specified area code in the Asterisk CLI. I don't think this is possible with the current AstDB code. > If this > were a SQL database it would be som

Re: [asterisk-users] AstDB wildard searches

Geoff Lane wrote: > asterisk -rx "database show cidname" | grep > at the Linux shell prompt, but I'm looking for a way to do this > without leaving the Asterisk CLI. > I don't believe it does. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temp

[asterisk-users] AstDB wildard searches

Hi All, I'm looking for a way to filter the AstDB cidname family to show only those entries with a specified area code in the Asterisk CLI. If this were a SQL database it would be something like: SELECT number, name FROM cidname WHERE number LIKE '1234%' I've tried "database show cidname 1234*" an

[asterisk-users] TeleKaam - Voice Portal for Students and Parents

Hello All We have started a voice portal for Parents and Students. They can listen Grades, Attendance status and other relevant information over *phone*. Please read features below and to listen Demo IVR call at *00 92 42 8315427. *Initially we are deploying it for a School and planning to spread

Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

2009/2/16 SIP > Grygoriy Dobrovolskyy wrote: > > > > > > 2009/2/13 Tzafrir Cohen > > > > > > On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: > > > > > > I've been involved with getting better data for running Asterisk on > > > the Amazo

Re: [asterisk-users] Michael Graves post

On Mon, 16 Feb 2009 16:05:48 +1300, Matt Riddell wrote: >On 10/02/2009 5:08 a.m., Michael Graves wrote: >> I unwittingly started this on Facebook, which I don't user very much. >> Here's the gist of it. >> >> A Strange Brew: VoIP/Telephony Crossed With Surround Sound >> >> It couldn't be the pur

Re: [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)

> > Sorry about off-topic, but can you advise the mail client who is > able to organise the web mailing list topic as web interface does ? > (i mean by blocks/topics) I wold be glad to use something else with > the same usability, but dont see any alternative. > Thank you Just turn on thread

Re: [asterisk-users] Faxing with asterisk

The Attrafax software that was mentioned at the beginning of the thread does support Gateway mode. Regards, Marc -Original Message- Fabio Mosti wrote: > 2009/2/16 Steve Underwood : > > >> You don't indicate the kind of setup you are using. >> > > I use asterisk (Spandsp) with a

Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

Grygoriy Dobrovolskyy wrote: > > > 2009/2/13 Tzafrir Cohen > > > On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: > > > > I've been involved with getting better data for running Asterisk on > > the Amazon EC2 cloud computing system. Here

Re: [asterisk-users] Faxing with asterisk

Fabio Mosti wrote: > 2009/2/16 Steve Underwood : > > >> You don't indicate the kind of setup you are using. >> > > I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to > another asterisk (zap). > > client->asterisk (Spandsp)->asterisk (zap)->fax To quote the Mythbusters, "t

Re: [asterisk-users] OpenSky: Digium Skype gateway?

2009/2/13 John Todd > > On Feb 13, 2009, at 11:19 AM, Philipp von Klitzing wrote: > > > Hi there, > > > > is gizmo the first user of the Digium Skype solution, or do they use a > > different approach/product - any clue? > > > > http://www.gizmo5.com/pc/opensky/ > > > > Philipp > > > I know nothin

Re: [asterisk-users] Faxing with asterisk

Fabio Mosti wrote: > I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to > another asterisk (zap). > > client->asterisk (Spandsp)->asterisk (zap)->fax > We have two remote phone systems connected via IAX and 1 fax server at the corporate offices with a PRI (ZAP). The fax serv

Re: [asterisk-users] Faxing with asterisk

2009/2/16 Steve Underwood : > You don't indicate the kind of setup you are using. I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to another asterisk (zap). client->asterisk (Spandsp)->asterisk (zap)->fax > Regards, > Steve Best Regards, Fabio > > >

[asterisk-users] SpanDSP question for Steve

Hello- Firstly thanks very much for the work you have put into SpanDSP and the time you spend to assist people here :-) I am currently running SpanDSP 0.0.5 with Call Weaver. Is there any or sufficient gain to be had from upgrading SpanDSP? Michael

Re: [asterisk-users] Faxing with asterisk

Steve Underwood wrote: > Fabio Mosti wrote: > >> Hi All, >> >> I need to setup asterisk to receive fax. >> >> I'm try Spandsp (opensource) and Attrafax (commercial) both on >> asterisk 1.4.23) but the results are disappointing. >> with spandsp many times the fax arrives cut. >> with Attrafax i h

Re: [asterisk-users] Faxing with asterisk

Fabio Mosti wrote: > Hi All, > > I need to setup asterisk to receive fax. > > I'm try Spandsp (opensource) and Attrafax (commercial) both on > asterisk 1.4.23) but the results are disappointing. > with spandsp many times the fax arrives cut. > with Attrafax i have some problem. > > Anyone have any

Re: [asterisk-users] Faxing with asterisk

2009/2/16 Grygoriy Dobrovolskyy > > > 2009/2/16 Michael > >> >> > > Anyone have any idea or solution (Opensource or commercial) to suggest >> me >> > > ? >> > > >> > > Best Regards >> > > >> > > Try hylafax with IAXmodem. The best results i had it the multitech >> modems >> > >> > directly conne

Re: [asterisk-users] Faxing with asterisk

2009/2/16 Michael > > > > Anyone have any idea or solution (Opensource or commercial) to suggest > me > > > ? > > > > > > Best Regards > > > > > > Try hylafax with IAXmodem. The best results i had it the multitech > modems > > > > directly connected to FXS PCI card, you have a nice web interface

Re: [asterisk-users] Faxing with asterisk

> > Anyone have any idea or solution (Opensource or commercial) to suggest me > > ? > > > > Best Regards > > > > Try hylafax with IAXmodem. The best results i had it the multitech modems > > directly connected to FXS PCI card, you have a nice web interface if you > wish also (avantfax) You can fin

Re: [asterisk-users] Faxing with asterisk

On Mon, 16 Feb 2009 22:14:49 Fabio Mosti wrote: > Hi All, > > I need to setup asterisk to receive fax. > > I'm try Spandsp (opensource) and Attrafax (commercial) both on > asterisk 1.4.23) but the results are disappointing. > with spandsp many times the fax arrives cut. > with Attrafax i have some

Re: [asterisk-users] Asterisk on EC2 cloud computing - price assumptions - your brain needed

2009/2/13 Tzafrir Cohen > On Fri, Feb 13, 2009 at 09:59:50AM -0800, John Todd wrote: > > > > I've been involved with getting better data for running Asterisk on > > the Amazon EC2 cloud computing system. Here are some calculations > > I've made on costs based on current published prices on Amazo

Re: [asterisk-users] Faxing with asterisk

2009/2/16 Fabio Mosti > Hi All, > > I need to setup asterisk to receive fax. > > I'm try Spandsp (opensource) and Attrafax (commercial) both on > asterisk 1.4.23) but the results are disappointing. > with spandsp many times the fax arrives cut. > with Attrafax i have some problem. > > Anyone have

Re: [asterisk-users] [OT] Gmail is broken (was: Re: WiFi SIP phone w/VPN?)

2009/2/13 Philipp Kempgen > Benny Amorsen schrieb: > > > Top posting is annoying. Gmail is broken; maybe I should just killfile > > @gmail.com. > > Emails sent through Gmail's *web interface* are broken. :-) > > > Philipp Kempgen > > -- > AMOOCON 2009, May 4-5, Rostock / Germany -> http://w

Re: [asterisk-users] No such command 'core stop now'

Can anyone help? On Sun, Feb 15, 2009 at 2:51 PM, Jim Boykin wrote: > It does not work at all even after long time. DNS resolution is not a > problem, because if I load it from command line "asterisk -c", > everything works fine. > > The problem is when it is configured to be loaded from /etc/ini

Re: [asterisk-users] Gizmo SIP / Skype gateway

On Monday, February 16, 2009, Julian Lyndon-Smith wrote: > We also don't yet know the pricing structure of chan_skype ... I thought it was $99 per channel for corporate licenses or $19 for a single, personal license ... or have I got the wrong ChanSkype? http://www.chanskype.com follow the "buy"

[asterisk-users] Faxing with asterisk

Hi All, I need to setup asterisk to receive fax. I'm try Spandsp (opensource) and Attrafax (commercial) both on asterisk 1.4.23) but the results are disappointing. with spandsp many times the fax arrives cut. with Attrafax i have some problem. Anyone have any idea or solution (Opensource or comm

Re: [asterisk-users] Gizmo SIP / Skype gateway

David Quinton wrote: > On Sun, 15 Feb 2009 15:01:42 +, Julian Lyndon-Smith > wrote: > > >> Anyone got any thoughts on this and how it compares to the chan_skype >> that's due soon ? >> >> "OpenSky is a free service provided by Gizmo5 which allows *any* mobile >> phone, web browser or IP a