[asterisk-users] Druid 2.0 released from the Druid Open Source Unified Communications Project

Dear Asterisk users, We would like to announce that Druid, Open Source Unified Communications project has just made a major release: Druid 2.0. It is out!It has a ton of new features and a highly improved interface. Asterisk stability has also been greatly improved. For more info http://forums.voi

Re: [asterisk-users] faxing via linksys SPA3102 half page goes through

On Wed, 04 Mar 2009 19:25:38 Joseph wrote: > I'm faxing from stand alone fax machine via linksys SPA3102 but most of > the time only half or quarter page goes through. > > Did anybody have any experience like this? Should be obvious but does your up line SIP provide support T.38? ___

[asterisk-users] faxing via linksys SPA3102 half page goes through

I'm faxing from stand alone fax machine via linksys SPA3102 but most of the time only half or quarter page goes through. Did anybody have any experience like this? -- #Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] Silk for Free

On Tue, Mar 03, 2009 at 11:51:56PM -0500, Dean Collins wrote: > http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio > _codec.html?tk=rss_news > > any thoughts? What patents does Skype have that are required for implementing this codec? -- Tzafrir Cohen icq#16

[asterisk-users] Silk for Free

http://www.pcworld.com/article/160653/skype_gives_away_highquality_audio _codec.html?tk=rss_news any thoughts? Regards, Dean Collins Cognation Inc d...@cognation.net +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial).

Re: [asterisk-users] building a phone

It may not be necessary to replace Snom's firmware to add interesting functionality to the product. Though that was not the original poster's premise, admittedly. As to the 'loose ends', they usually remain so until someone is motivated to drive them to closure. Absence of a suitable hardware

Re: [asterisk-users] $20 Bounty

On Tue, Mar 3, 2009 at 10:25 PM, David Backeberg wrote: exten => 123,s,1 Playback(enterzipcode) exten => 123,s,n Read(zip||5) exten => 123,s,n System(wget http://pathtoyahooservice${zip} -o forecast.txt) exten => 123,s,n System(wget --post-file forecast.txt -o wav.url) exten => 123,s,n System(wge

[asterisk-users] [SPAM] RE: $20 Bounty

Dear Sir: Me no peak englisss…. 20 pesos??? ok , tank you sir, tank you sir From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kinjal Dixit Sent: Tuesday, March 03, 2009 9:57 PM To: Asterisk Users Mailing List -

Re: [asterisk-users] $20 Bounty

On Tue, Mar 3, 2009 at 7:12 PM, Dean Collins wrote: > I’ll pay anyone a $20 bounty for someone to replicate the USA Asterisk > Weather App on Tropo. All you have to do is violate the ToS on a few services: wget the weather from yahoo, for instance: http://weather.yahooapis.com/forecastrss?p=06513

Re: [asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

Use something like this: exten => 100,1,Dial(SIP/100) exten => 1001,1,Set(CALLER=${CALLERID(NUM)}) exten => 1001,2,Dial(SIP/100) exten => 1001,3,Goto(default,${CALLER}) Brandon B. On Tue, Mar 3, 2009 at 8:06 PM, James Mutuku wrote: > Hellos, > > I want to configure asterisk so that if exten A

[asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

Hellos, I want to configure asterisk so that if exten A transfers a call to exten B, and B is either busy or the call is not answered, the call returns back to A. Is this possible? Please help James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department ad

Re: [asterisk-users] $20 Bounty

> > I wish my family and I could "live" on $40 a week... > > simplify, simplify, simplify -- http://www.linkedin.com/in/kinjaldixit open networker ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To

Re: [asterisk-users] $20 Bounty

> >>I'd be willing to wager someone another $20 that he does have a > >> "taker" on that bounty within the next 48 hours. For those of us that > >> reside in countries where a single cup of coffee can run more than 20% > >> of that complete bounty, we scoff at the opportunity. For others, thei

Re: [asterisk-users] $20 Bounty

On Tue, 3 Mar 2009, BJ Weschke wrote: > I'd be willing to wager someone another $20 that he does have a "taker" > on that bounty within the next 48 hours. For those of us that reside in > countries where a single cup of coffee can run more than 20% of that > complete bounty, we scoff at the opp

Re: [asterisk-users] $20 Bounty

Michael wrote: >>I'd be willing to wager someone another $20 that he does have a >> "taker" on that bounty within the next 48 hours. For those of us that >> reside in countries where a single cup of coffee can run more than 20% >> of that complete bounty, we scoff at the opportunity. For others

Re: [asterisk-users] $20 Bounty

>I'd be willing to wager someone another $20 that he does have a > "taker" on that bounty within the next 48 hours. For those of us that > reside in countries where a single cup of coffee can run more than 20% > of that complete bounty, we scoff at the opportunity. For others, their > family m

Re: [asterisk-users] $20 Bounty

OCG Technical Support wrote: > Perhaps if he threw in a paperclip and some tictacs people would respond... > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards > Sent: March 3, 2009 7:37 PM > To:

Re: [asterisk-users] $20 Bounty

Perhaps if he threw in a paperclip and some tictacs people would respond... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: March 3, 2009 7:37 PM To: Asterisk Users List Subject: Re: [asteris

Re: [asterisk-users] after install the zaptel but the rtp failed

hi Grygoriy : appreciate your reply , that's my cli command: CLI> zap show status Description Alarms IRQbpviol CRC4 ZTDUMMY/1 1 UNCONFIGUR 0 0 0 Is't all right? forward your echo . thanks 2009

Re: [asterisk-users] $20 Bounty

On Tue, 3 Mar 2009, Dean Collins wrote: > I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk > Weather App on Tropo. Wow. $20. :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice:

[asterisk-users] AEL2: If-then-else not permitted in Switch-Case

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted I

[asterisk-users] $20 Bounty

http://saunderslog.com/2009/03/03/voxeo-launches-tropocom-mashup-platfor m/ I'll pay anyone a $20 bounty for someone to replicate the USA Asterisk Weather App on Tropo. Would like to see how quickly this is implemented. Regards, Dean Collins Cognation Inc d...@cognation.net

Re: [asterisk-users] Predefined viables

On Wed, 4 Mar 2009, michel freiha wrote: > Does anyone knows how to add a new variable to the predefined variables > sent by asterisk to AGI script? You don't. If you really wanted to add to the AGI environment variables, you could hack on res_agi.c, but I don't think this is what you want to

[asterisk-users] Predefined viables

Hi all, Does anyone knows how to add a new variable to the predefined variables sent by asterisk to AGI script? regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update opti

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

On 22/02/2009 6:35 a.m., Asterisk Asterisk wrote: > If anyone needs the data output by this module (test data for the past few > days), I'd be happy to share. It includes the date/time, gender detected, > whether that's correct according to the tester, the winning ratio, and the > energy levels

Re: [asterisk-users] Asterisk analog DID with Adit 600

Is it possible to set the Executone to NOT wink? Without the wink, a simple FXO port will DP into the switch without problems and without using E&M signaling Even an X100 card could do the job ( at least with earlier Asterisk ) . Am I reading that late Asterisk has mucked up the driver for this o

[asterisk-users] Unable to create channel

Dear All, I have the following warnings on my log file: [Mar 3 15:26:23] NOTICE[22639] chan_local.c: No such extension/context 0.0@default creating local channel [Mar 3 15:26:23] WARNING[22639] app_dial.c: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) [Mar 3 15:26:24] WARNING

[asterisk-users] SOLVED - Re: Asterisk analog DID with Adit 600

Dave Fullerton wrote: > Hello All, > > I'm trying to connect Asterisk to an Executone phone system with an > analog DID card and I'm hoping someone can help me figure out what I'm > doing wrong. The Executone DID card provides battery to the telco, when > the telco wishes to dial a DID it goes

[asterisk-users] monitoring a channel and redirect to conf

I am monitoring a channel... I then redirect that channel to a conf with lq as options. When playing back the gsm file I have all recording upto the point of redirect to the conference. How do I CONTINUE to record and not loose anything after redirecting to the conference? After redirecting I h

Re: [asterisk-users] CDR

On Tue, 2009-03-03 at 09:24 -0300, Gustavo A Gonzalez wrote: > Hello, Is there a field into the Zapata.conf file that affect the > value for the fields “start” and “answer” in the CDR? I have this > fields with the same value and I dont understand why. Thanks!! This is because analog inte

[asterisk-users] Asterisk analog DID with Adit 600

Hello All, I'm trying to connect Asterisk to an Executone phone system with an analog DID card and I'm hoping someone can help me figure out what I'm doing wrong. The Executone DID card provides battery to the telco, when the telco wishes to dial a DID it goes off-hook, waits for a wink from

Re: [asterisk-users] clone X100p+dahdi dial out works only after receiving call

On Tue, 3 Mar 2009 12:04:48 +0200 Tzafrir Cohen wrote: > http://bugs.digium.com/view.php?id=14577 ? Totally. Switching to fxsls from fxsks did the trick. Do I know what difference this makes to me? Not at all... but it 'fixes' the problem CHANUNAVAIL when it *should* be good to go. http://bu

Re: [asterisk-users] macro-stdexten question

On Tue, 3 Mar 2009, Danny Nicholas top posted a seemingly unrelated sequence of words: > Macro-stdexten is not explicitly called. It is a "drop through" > function of Dial. Look at voip.org for better details. I thought since the name started with "macro" that it was a macro, not a function.

Re: [asterisk-users] macro-stdexten question

Macro-stdexten is not explicitly called. It is a "drop through" function of Dial. Look at voip.org for better details. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Tuesday, March 03, 200

Re: [asterisk-users] benchg729 - no valid g729 license

Kevin P. Fleming escreveu: > Leonardo Gomes Figueira wrote: > >> Do I have to contact Digium to reset the license key even if I'm not >> trying to register a new MAC ? I just want an updated license key file. > > I'd like to help you debug this, so we can improve our registration system. > > Can

Re: [asterisk-users] macro-stdexten question

> The trace showed: > > -- Executing [6...@dlpn_defaultdialplan:1] > Macro("IAX2/6223-10489","stdexten|6123|SIP/6123&IAX2/6123") in new > stack > > Which means that this channel was in context DLPN_DefaultDialPlan , > extension 6123 in the dialplan . > > Next step: > > dialplan show 6...@dlpn

Re: [asterisk-users] macro-stdexten question

On Tue, Mar 03, 2009 at 07:41:13AM -0800, Haim Dimer wrote: > Thanks Tzafrir, but my questions remains :-) Who's calling this macro > if all I can find in the /etc/asterisk files in the definition of the > macro, not any part of the dialplan actually calling it. Here, check > this out: > > r

Re: [asterisk-users] early dial (or overlap dial) and Asterisk 1.2 vs. 1.4

Just in case someone can warn me of possible asterisk breakage, I'm attaching a patch I just applied to chan_sip for * 1.2.31.1. It's been working fine for now with "urlencode=yes" in sip.conf. asterisk-1.2.31-chan_sip.diff Description: Binary data _

Re: [asterisk-users] macro-stdexten question

Thanks Tzafrir, but my questions remains :-) Who's calling this macro if all I can find in the /etc/asterisk files in the definition of the macro, not any part of the dialplan actually calling it. Here, check this out: r...@asterisk-server:/etc/asterisk# grep stdexten * extensions.conf:[macr

Re: [asterisk-users] macro-stdexten question

It is "sort of" hard-coded. Do the grep for stdexten instead of "macro-stdexten" (* adds the macro- at runtime). The Dial command calls/drops into it as part of fallthrough. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com]

Re: [asterisk-users] macro-stdexten question

On Tue, Mar 03, 2009 at 07:28:51AM -0800, Haim Dimer wrote: > I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. > > When one phone calls another, I see the following on the console > (here, 6223 dials 6123) > > -- Executing [6...@dlpn_defaultdialplan:1] Macro("IAX2/6223-10489",

Re: [asterisk-users] AGI problem using mono (.Net)

Hi douglas, I made an AGI script with curl in my dial-plan. Take a look: [custom-login] exten => s,1,Set(LOGINOK=0) exten => s,2,Read(codigo|custom/codigo|4||3|3) exten => s,3,Read(clave|custom/clave|4||3|3) exten => s,4,Set(LOGINOK=${CURL(http:///cgi-bin/login.pl?codigo=${codigo}&clave=${clave

[asterisk-users] macro-stdexten question

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6...@dlpn_defaultdialplan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [...@ma

Re: [asterisk-users] Remote Connection to Asterisk

So how do you want to make your calls? You state that you have a TDM400 but no POTS lines installed. To make a ZAP/DAHDI call you have to put a line to the TDM400. If you're going to make your calls via SIP, the TDM400 serves no purpose. Show us your Dialplan (the part that runs the Dial comman

Re: [asterisk-users] change language and playback issue

2009/3/3 Philipp Kempgen > Giedrius Augys schrieb: > > Only need to make change in file.c : > > int ast_language_is_prefix = 0 > > And that's all! > > > > 2009/2/27 Giedrius Augys > >> I have problem with Asterisk 1.6.0.1. I need to change language for > >> playing prompts in Lithuanian. But i

[asterisk-users] Remote Connection to Asterisk

Hello all - This is basically an updated re-posting of one I've posted a few days ago. Thanks to the kind help provided but I still can't make it work. But I'm moving a little further down the line (thanks to you folks). Basically, I've got an Asterisk server in a LAB ENVIRONMENT on my home LAN

[asterisk-users] [OT] "please help" (was: Re: incoming call problem)

David fire schrieb: > you should post it again whit a subject like "T38 problem" or "please > help!! t38 problem". Right. You should definitely add more exclamation marks and write all uppercase. Examples: "URGENT! Please help!!!" "ASTERISK PROBLEM!!" ;-) Don't! Philipp Kempgen -

Re: [asterisk-users] Dialing with cli

You can use console dial num...@context. If the Asterisk box has a soundcard, you will hear the audio and will be able to speak on the microphone. Vinícius Fontes www.asteriskforum.com.br - "Joseph L. Casale" escreveu: > Any way to initiate a call and execute a playback of an audio file

Re: [asterisk-users] How to set PRI line timeout value

Jim Dickenson schrieb: >> > From: "Kevin P. Fleming" >> > Organization: Digium, Inc. >> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion >> > >> > Date: Mon, 02 Mar 2009 14:41:51 -0600 >> > To: Asterisk Users Mailing List - Non-Commercial Discussion >> > >> > Subject: Re: [ast

Re: [asterisk-users] change language and playback issue

Giedrius Augys schrieb: > Only need to make change in file.c : > int ast_language_is_prefix = 0 > And that's all! > > 2009/2/27 Giedrius Augys >> I have problem with Asterisk 1.6.0.1. I need to change language for >> playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime >> pl

[asterisk-users] patlooptest and TE121P

Hi List. I'm running the patlooptest program I've found in dahdi_tools 2.1.0.2. The target is a TE121P board with a loopback cable inserted on the socket. I suppose that the loopback is working fine because I'm able to see the green led on and dahdi_tool reports no errors. When I run the patloopte

Re: [asterisk-users] tons of open SIP channel between two snom 360

2009/3/3 Giorgio Incantalupo : > Hi, > > I'm monitoring an Asterisk 1.2.18 box because sometimes I get two Snom > 360 phones creating a lot of SIP channels between them and it seems they > never die. > How can it be? > > Thank you. > > Giorgio I would suggest looking for network dropouts and for p

[asterisk-users] NOTIFY/SUBSCRIBE and MWI in 1.4

Hi, >From recent experiences with 1.4.23.1, Asterisk doesn't need any incoming MWI-related SUBSCRIBE message to send NOTIFY messages changing phones MWI status. This is fine for me but I'm wondering what if I were using SIP hardphones refusing any such NOTIFY without prior SUBSCRIBE (does such pho

Re: [asterisk-users] Master.csv missing dstchannels values

On 3 Mar 2009, at 11:59, Tobias Steen wrote: > In some rows there are missing values for dstchannel where lastapp > is “Queue” and lastdata equals the queue name I’m interested in. > > Pleas guide me, are the rows including the missing dstchannel values > equal to calls hanged up by the caller

[asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy

Hellos, I want to configure asterisk so that if exten A transfers a call to exten B, and B is either busy or the call is not answered, the call returns back to A. Is this possible? Please help James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:

[asterisk-users] Master.csv missing dstchannels values

Hi, I'm currently building an report query for Asterisk (based on the Master.csv file). In some rows there are missing values for dstchannel where lastapp is "Queue" and lastdata equals the queue name I'm interested in. Pleas guide me, are the rows including the missing dstchannel values equal

[asterisk-users] cdr database

Dear sir or madam, I am working on a project which uses asterisk server. In cdr event which is triggered after every call, there are two records called "start time" and "end time" but these records are not getting saved in the cdr database. after I looked at the codes I found that by default these

[asterisk-users] CDR

Hello, Is there a field into the Zapata.conf file that affect the value for the fields “start” and “answer” in the CDR? I have this fields with the same value and I dont understand why. Thanks!! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. ggonza...@despegar.com

Re: [asterisk-users] after install the zaptel but the rtp failed

2009/3/3 邱磊 > hi everyone: > now ,i have a strange situation: I want to make a meetme conference and > install the zaptel1.4* in my asterisk. > every things seem well but it did't work normally. > I use the Playback app for test .It didn't reply any voice.I tried in > another asterisk server the

[asterisk-users] after install the zaptel but the rtp failed

hi everyone: now ,i have a strange situation: I want to make a meetme conference and install the zaptel1.4* in my asterisk. every things seem well but it did't work normally. I use the Playback app for test .It didn't reply any voice.I tried in another asterisk server the playback app work well.

Re: [asterisk-users] Dialing with cli

On Mon, Mar 02, 2009 at 04:13:58PM -0700, Joseph L. Casale wrote: > Any way to initiate a call and execute a playback of an audio file from > the cli? The Asterisk CLI? 'originate' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406

Re: [asterisk-users] clone X100p+dahdi dial out works only after receiving call

On Sat, Feb 28, 2009 at 11:24:53AM -0800, Michael Higgins wrote: > > So, tweaking configs, rebuilding this and that... restarting, twiddling, it > works (yeah!), but fails on re-boot to work at all. Consistently, though. > > I believe it comes down to this: I can call out only *after* I've recei

Re: [asterisk-users] Access sip.conf's mailbox from dialplan ?

2009/3/3 Olivier > Hello, > > In sip.conf, each peer/friend/user entry gathers several parameters such as > type, canreinvite or mailbox. > How can you specifically access to mailbox value from dialplan ? > > I know how to access custom parameters (ie setvar=FOO=value) but I don't > know to acces

[asterisk-users] Access sip.conf's mailbox from dialplan ?

Hello, In sip.conf, each peer/friend/user entry gathers several parameters such as type, canreinvite or mailbox. How can you specifically access to mailbox value from dialplan ? I know how to access custom parameters (ie setvar=FOO=value) but I don't know to access standard parameters. I'm speci

[asterisk-users] tons of open SIP channel between two snom 360

Hi, I'm monitoring an Asterisk 1.2.18 box because sometimes I get two Snom 360 phones creating a lot of SIP channels between them and it seems they never die. How can it be? Thank you. Giorgio A "show channels" excerpt follows: SIP/20-08a7aa80 (None) Up Bridged Call(