Re: [asterisk-users] SLA broken in 1.4.3?

2007-05-05 Thread Jay Austad
Doh, I didn't compile meetme support into this install because it said it required a zaptel card for the hardware timers. I have a spare zaptel card, but I'd rather not install it if I don't have to. I didn't know of the ztdummy module, I will compile this and then recompile asterisk. I

RE: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?

2007-05-05 Thread Salvatore Giudice
My money is on compulsory drug rehab or simply being held for 45 days of observation after being caught sexually abusing a pony. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Bl

RE: [asterisk-users] asterisk telemarketer torture sound files

2007-05-05 Thread Salvatore Giudice
Just forward them to 1-800-big-dick or some other 800 toll free phone sex line. They can't tell they've been forwarded. They'll figure it out eventually. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.

RE: [asterisk-users] Asterisk x legacy pabx

2007-05-05 Thread Salvatore Giudice
It’s basically the same problem. Asterisk is not a standalone voicemail server. It would have to support Qsig. Asterisk doea not exactly have expansive Qsig support. I believe there are several bounties out for Qsig. Without Qsig, you would have to use parallel forking and ring the user’s avay

Re: [asterisk-users] ASA-2007-013: IAX2 users can cause unauthorizeddata disclosure

2007-05-05 Thread Dovid B
Has 1.2.19 been released ? - Original Message - From: "Kevin P. Fleming" <[EMAIL PROTECTED]> To: Sent: Friday, May 04, 2007 12:20 PM Subject: [asterisk-users] ASA-2007-013: IAX2 users can cause unauthorizeddata disclosure Asterisk Project Security Advisory - AS

Re: [asterisk-users] question about more than one drop file

2007-05-05 Thread dave cantera
shawn, you can set an archive variable in the .call file to 'yes' and it will save it in ./outgoing_done... if there is now outbound line availible, the .call file is updated (appended to) as per the status... * will keep trying till it completes the calls or the number of retries is reached.

Re: [asterisk-users] GXV-3000 IP Video Phone

2007-05-05 Thread dave cantera
nitesh, you are correct. you need 1.4.x... daveC Nitesh Divecha wrote: Hello All, I just received some test units of Grandstream GXV-3000 IP Video Phone. I did some research and looks like Asterisk 1.2 does not support video H.264 but Asterisk 1.4 does. Is it correct? Actually I did try to

[asterisk-users] auto call out via drop file ERROR: 'OutgoingSpoolFailed'

2007-05-05 Thread dave cantera
has anyone run into this message? for some reason, which I can not determine, this script stop working and now gives this error. I googled 'outgoingspoolfailed' but not too much turned up... only questions, no answers... :( I am mv'ng a .call file to the ./outgoing directory. the call init

RE: [asterisk-users] RXFAX/TXFAX

2007-05-05 Thread Michelle Dupuis
SUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Informacin de NOD32, revisin 2243 (20070505) __ Este mensaje ha sido analizado con NOD32 antivirus system http://www.nod32.com ___ --Bandw

Re: [asterisk-users] using Playback() to play a random sound file

2007-05-05 Thread dave cantera
steve, thats Great... my C is old and ftw operated differently on sysV, solaris, sunos, ultrix, and osf... so I went back to bourne... couldn't work through the idiosyncracies of gnu autoconf, etc... although I have a many reasons to, I just couldn't get to production 'C' coding level... d

[asterisk-users] Channel / Exten Status

2007-05-05 Thread Pablo L. Arturi
Hello guys, I am using the API Mananger with PHP to initiate a call from a webpage. First I call to a line number, and then to an asterisk extension. I followed examples on using the API Mananger, without any problem, and working great. Now I have a problem. I can initiate the call. I can call

[asterisk-users] Dial Plan for Multi-Location & Support Queue

2007-05-05 Thread Deepak Naidu
Hi, I am in the process of planning a dial plan, In regards to the requirement, I am confused how to go about the dial plan. The scenario is like below. BRANCH - A - (COMPANY) Line 1 -- Extension 239 Line 2 -- Extension 8239 BRANCH - B - (COMPANY) Lin

RE: [asterisk-users] OT: USB T1/E1 Interface?

2007-05-05 Thread Michael Collins
To everybody: Thanks for your thoughts and suggestions. This will be my last post to this list on this subject. I've started a blog about my research into this project: http://myossjourneys.blogspot.com/ If you want to discuss this any further please do so over there. Thanks again! -MC >

[asterisk-users] ${ANSWEREDTIME} Broken on 1.2.13?

2007-05-05 Thread Barton Fisher
No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58

Re: [asterisk-users] TDM400P usada?

2007-05-05 Thread Carlos Rojas
Hey Look http://www.asterisk-es.org Best Regards On 5/5/07, Cesar Benjamin Garcia Martinez <[EMAIL PROTECTED]> wrote: Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de mexico, asi que en parte tienes razón, pero tb creo que deberías haber puesto de donde eres. *De

[asterisk-users] res_config_pgsql.c in * 1.4.4

2007-05-05 Thread Gavin Henry
Dear All, Where can I find a res_pgsql.conf and some sql to insert for tables etc.? Are all db res things to be done via odbc now? Why was this included with no docs or sample conf? Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.

[asterisk-users] I'm looking for solution

2007-05-05 Thread Ardit Saliu
HI I have 3 Linksys SIP901 IP phones I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd I'm looking to connect this phones together and to make calls between them Not from outside of my lan I don't know how to configure asterisknow beta Can somebody help I'm d

Re: [asterisk-users] I'm looking for solution

2007-05-05 Thread Remco Post
Ardit Saliu wrote: > HI > > I have 3 Linksys SIP901 IP phones > > I also have a pc I’m not using it amd athlon 1800+ 512mb ram and 40 gb hdd > > I’m looking to connect this phones together and to make calls between them > > Not from outside of my lan > > > > I don’t know how to configure as

Re: [asterisk-users] asterisk telemarketer torture sound files

2007-05-05 Thread Dave Miller
Adam Jacob Muller wrote on 5/5/07 1:38 PM: > > On May 5, 2007, at 1:15 PM, Dave Miller wrote: > >> Adam Jacob Muller wrote on 5/5/07 1:06 PM: >>> Hi, >>> I have some annoying telemarketer calling me on a recurring basis, but >>> I'd like to discourage them a bit and have some fun. >>> I found thi

[asterisk-users] I'm looking for solution

2007-05-05 Thread Ardit Saliu
HI I have 3 Linksys SIP901 IP phones I also have a pc I'm not using it amd athlon 1800+ 512mb ram and 40 gb hdd I'm looking to connect this phones together and to make calls between them Not from outside of my lan I don't know how to configure asterisknow beta Can somebody help I'm d

Re: [asterisk-users] AsteriskNow!

2007-05-05 Thread Steve Edwards
On Sat, 5 May 2007, Bill Merriam wrote: To get root access type "sudo su". Once there you can change the root password with "passwd". Seems a bit redundant: -fs::sedwards:~$ man sudo NAME sudo - execute a command as another user -fs::sedwards:~$ man su

Re: [asterisk-users] RXFAX/TXFAX

2007-05-05 Thread Andreas van dem Helge
On 5/4/07, Cesar Benjamin Garcia Martinez <[EMAIL PROTECTED]> wrote: Hi all Somebody can tell me, what way i can send/receive faxes with asterisk 1.4??? In Asterisk 1.2.x i have use the spandsp lib and app_rxfax / app_txfax from soft-switch.org for add T.38 support to asterisk and to have that

[asterisk-users] Asterisk 1.4.4 and Custom Postgres 8.2.4 (checking for PQexec in -lpq... no)

2007-05-05 Thread Gavin Henry
Dear All, Why does my configure fail like so: checking for pg_config... /usr/local/pgsql/8.2.4/bin/pg_config checking for PQexec in -lpq... no configure: *** configure: *** The PostgreSQL installation on this system appears to be broken. configure: *** Either correct the installation, or run con

Re: [asterisk-users] Re: Anyone having trouble with claling US Domesticon Sellvoip?

2007-05-05 Thread Tom Lynn
At the very least, he's abusing his customers. Substances? I hadn't thought of that. On 4/30/07, Salvatore Giudice <[EMAIL PROTECTED]> wrote: I suspect that Jed has a substance abuse problem and that he may be in rehab. I don't know for sure of course. This kind of silence is indicative of pe

Re: [asterisk-users] RXFAX/TXFAX

2007-05-05 Thread Tzafrir Cohen
On Sat, May 05, 2007 at 02:25:50PM -0500, Cesar Benjamin Garcia Martinez wrote: > My question is becouse i read than 1.4 supports T.38, and then should > receive fax i guess... 1.4 only supports VoIP passthrough of T.38 . That is: if you get a T.38 fax, it can be safely redirected to a T.38-capa

[asterisk-users] ODBC

2007-05-05 Thread Forrest Beck
I am trying to compile asterisk with ODBC support on CentOS 4.4. I am running into the same issue as documented in this bug. http://bugs.digium.com/view.php?id=8214 The server is a Dell 2950 with Dual Core /64bit processors (2Gig RAM). I tried creating a symbolic link link mentioned in the bug

Re: [asterisk-users] AsteriskNow!

2007-05-05 Thread Bill Merriam
Ed Nuñez wrote: > > > Does anyone know how to gain access directly to the configuration files > in AsteriskNow? I have dual NICs and need to change the binding in the > config file. I believe they blocked ssh2 access by default. > ssh is not blocked. You have to ssh into the userid admin.

RE: [asterisk-users] TDM400P usada?

2007-05-05 Thread Cesar Benjamin Garcia Martinez
Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de mexico, asi que en parte tienes razón, pero tb creo que deberías haber puesto de donde eres. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Rodrigo Mercado Enviado el: sábado, 05 de mayo de 2007 12:38 Para:

RE: [asterisk-users] RXFAX/TXFAX

2007-05-05 Thread Cesar Benjamin Garcia Martinez
rmacin de NOD32, revisin 2243 (20070505) __ Este mensaje ha sido analizado con NOD32 antivirus system http://www.nod32.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update opti

RE: [asterisk-users] Asterisk-Polycom HELLLLPPP!!!!

2007-05-05 Thread Jim Suber
Thanks I did that as well. I did however get the problem fixed by setting canreinvote=yes Apparently the polycom wants it when the soft phones don't Sorry, I meant canreinvite can re inVOTE- is something that dead people do in my home state of Mississippi. -Original Message- From: [EMAIL

Re: [asterisk-users] Asterisk x legacy pabx

2007-05-05 Thread Josué Conti
Hi Salvatore, thanks for reply. And if pabx legacy was Siemens model HiPath 3750, could use asterisk as serving of voicemail and other applications? Best Regards Josué 2007/5/5, Salvatore Giudice <[EMAIL PROTECTED]>: I've done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Q

[asterisk-users] GXV-3000 IP Video Phone

2007-05-05 Thread Nitesh Divecha
Hello All, I just received some test units of Grandstream GXV-3000 IP Video Phone. I did some research and looks like Asterisk 1.2 does not support video H.264 but Asterisk 1.4 does. Is it correct? Actually I did try to test with Asterisk 1.2 and video did not initialize but voice worked...

RE: [asterisk-users] SLA broken in 1.4.3?

2007-05-05 Thread David W. Rice
SLA requires meetme which requires at a minimum ztdummy. So, you must compile and install zaptel, then compile and install asterisk 1.4.3 and the sla commands will be in the CLI. Let me know if you need help setting up SLA on Polycom phones with *. I've done it successfully and have the configs.

Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04

2007-05-05 Thread Remco Post
Tzafrir Cohen wrote: > On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote: >> Mark Coccimiglio wrote: >>> Tzafrir; >>>Actually I have found this config to work really well. I prefer to >>> use a script run from inittab but Ubuntu doesn't work like Redhat or >>> BSD. On a production bo

Re: [asterisk-users] asterisk telemarketer torture sound files

2007-05-05 Thread Adam Jacob Muller
On May 5, 2007, at 1:15 PM, Dave Miller wrote: Adam Jacob Muller wrote on 5/5/07 1:06 PM: Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarkete

Re: [asterisk-users] TDM400P usada?

2007-05-05 Thread Rodrigo Mercado
Chile. No hay listas en español, y si lo enviè en español es justamente porque si alguien no lo habla no puede estar en CHILE, de todas formas muchas gracias por la amabilidad de traducir mi correo. saludos, bye bye On 5/5/07, Tom Rymes <[EMAIL PROTECTED]> wrote: On May 5, 2007, at 12:0

Re: [asterisk-users] Queue and voicemail

2007-05-05 Thread 0xception
You can add another line like exten=>0,1,VoiceMail([EMAIL PROTECTED]) this will catch the dialing of 0 before or after it enters the queue... but if you want them to be able to do that while in the queue then you need to add to your queue config a line like context=your-queue-context and then c

Re: [asterisk-users] asterisk telemarketer torture sound files

2007-05-05 Thread Dave Miller
Adam Jacob Muller wrote on 5/5/07 1:06 PM: > Hi, > I have some annoying telemarketer calling me on a recurring basis, but > I'd like to discourage them a bit and have some fun. > I found this: > http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture > but the link to download the soun

[asterisk-users] asterisk telemarketer torture sound files

2007-05-05 Thread Adam Jacob Muller
Hi, I have some annoying telemarketer calling me on a recurring basis, but I'd like to discourage them a bit and have some fun. I found this: http://www.voip-info.org/wiki/view/Asterisk+AEL+Telemarketer+Torture but the link to download the sound files is dead (wyoming.e-tools.com is NXDOMAIN)

Re: [asterisk-users] TDM400P usada?

2007-05-05 Thread Tom Rymes
On May 5, 2007, at 12:06 PM, Rodrigo Mercado wrote: Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. For anyone who is not a Spanish speaker, Rodrigo is looking for a used TDM400P card with FXS modules. He

Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04

2007-05-05 Thread Tzafrir Cohen
On Sat, May 05, 2007 at 06:23:43PM +0200, Remco Post wrote: > Mark Coccimiglio wrote: > > Tzafrir; > >Actually I have found this config to work really well. I prefer to > > use a script run from inittab but Ubuntu doesn't work like Redhat or > > BSD. On a production box keeping asterisk up an

[asterisk-users] Re: Virtual IP Adresses and SIP requests failing...

2007-05-05 Thread Christopher Aloi
Hello - Well I've been able to find a bit more about my problem. Again - I am not bound to a specific interface (0.0.0.0) When a SIP invite addressed to the .36 address, Asterisk replies FROM the .38 address. Is this the expected behavior? Wouldn't it make sense for Asterisk to reply on FROM

Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04

2007-05-05 Thread Remco Post
Mark Coccimiglio wrote: > Tzafrir; >Actually I have found this config to work really well. I prefer to > use a script run from inittab but Ubuntu doesn't work like Redhat or > BSD. On a production box keeping asterisk up and running is "THE TOP" > priority. If you would rather check every fi

[asterisk-users] TDM400P usada?

2007-05-05 Thread Rodrigo Mercado
Alguien tiene una TDM400P con modulo FXS usada a la venta ??, obviamente a precio de tarjeta usada... saludos, Rodrigo Mercado S. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

RE: [asterisk-users] Asterisk x legacy pabx

2007-05-05 Thread Salvatore Giudice
I’ve done a lot of work with Avaya. Voicemail systems attaché dot Avaya use Qsig trunk to pass calls to voicemail servers. The core of their modular messaging/message networking infrastructure can also use VPIM for communication between vmail servers. As far as I know, you can’t use Asterisk in the

Re: [asterisk-users] Manager API Output

2007-05-05 Thread Gordon Henderson
On Sat, 5 May 2007, Arun Kumar wrote: Hi, Is there any way that I can store my manager API output that is: Read The Fine WiKi!!! http://www.voip-info.org/wiki/index.php?page=Asterisk+manager+Example%3A+PHP Gordon ___ --Bandwidth and Colocation pro

Re: [asterisk-users] Asterisk x legacy pabx

2007-05-05 Thread Josué Conti
Hi Dean, thank you will be this attention. Currently asterisk is interconnected in pabx legacy through a A104D with protocol ISDN Qsig, uses LCR for routes of lesser cost and calls for other localities. But I see that domains of asterisk is limitless therefore would like to use it as also voicemai

Re: [asterisk-users] Manager API Output

2007-05-05 Thread Philipp Kempgen
Arun Kumar wrote: > Is there any way that I can store my manager API output that is: > My question is that is there any why using that I can get the QueueStatus > and store the result in some text file for further processing. > > > $strHost = "127.0.0.1"; >

Re: [asterisk-users] Queue and voicemail

2007-05-05 Thread Josué Conti
Hi all, thanks for this reply. Follows below the current configurations of mine asterisk, where the line functions perfectly, but does not obtain to rotear in case that no agent takes care of, for the voicemail. How it could give an option to the caller so that it can send a message? Sample: it di

[asterisk-users] SIP registration problem

2007-05-05 Thread Michelle Dupuis
I've reposted with a more meaningful subject - hopefully someone will replyWe have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attem

[asterisk-users] Manager API Output

2007-05-05 Thread Arun Kumar
Hi, Is there any way that I can store my manager API output that is: My question is that is there any why using that I can get the QueueStatus and store the result in some text file for further processing. thanks arun ___ --Bandwidth and Colocation

RE: [asterisk-users] Asterisk x legacy pabx

2007-05-05 Thread Dean Collins
Hi Josue, Yes you can use Asterisk along side an existing PABX. So with your existing Avaya you can allow it to connect to the handsets, but when calls are received for voicemail then you can send them to the Asterisk server another functionality you might find useful are conference rooms

[asterisk-users] TLS support

2007-05-05 Thread Alexandr Olekhnovich
Hello, Does anybody know whether Asterisk 1.4 supports TLS? Or may be any work patches or branches? Thanks in advance -- Best Regards Alexander Olekhnovich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUB

[asterisk-users] Queue Status

2007-05-05 Thread Arun Kumar
Hi I've few queues configured in * box is there any what that before sending call to a particular queue can we get the status of the queue that is how many agents are available in this queue (logged in, paused, busy, unavailable). thanks arun ___ --

Re: [asterisk-users] CDR(accountcode) empty in * 1.4.4 (for local chan)

2007-05-05 Thread Rizwan Hisham
Steve, I didnt mean to say that your patch did that. Actually i did saw this error before applying your patch. i just mentioned it here. So is this problem fixable? On 5/5/07, Steve Murphy <[EMAIL PROTECTED]> wrote: On Fri, 2007-05-04 at 15:25 +0500, Rizwan Hisham wrote: > Nops. removing res_fe

Re: [asterisk-users] Starting Asterisk on Ubuntu 7.04

2007-05-05 Thread Mark Coccimiglio
Tzafrir; Actually I have found this config to work really well. I prefer to use a script run from inittab but Ubuntu doesn't work like Redhat or BSD. On a production box keeping asterisk up and running is "THE TOP" priority. If you would rather check every five minutes then replace the fi

Re: [asterisk-users] RXFAX/TXFAX

2007-05-05 Thread Per Jessen
Cesar Benjamin Garcia Martinez wrote: > Somebody can tell me, what way i can send/receive faxes with asterisk > 1.4??? [snip] > How to i can send/receive fax to/from PSTN on asterisk 1.4 ? Check out a very recent thread on just that subject. Or go study how to use iaxmodem and hylafax. /Per J

Re: [asterisk-users] Queue and voicemail

2007-05-05 Thread 0xception
You could alternatively set a context for your queue in your config and create an extension for voicemail, if you would rather give the option to go to voice mail to the caller... (example: They can dial 0 to leave a message) On 5/4/07, Per Jessen <[EMAIL PROTECTED]> wrote: Josué Conti wrote:

Re: [asterisk-users] IP Phone Provisioning Tool by voip.com.sg - xml generation

2007-05-05 Thread Per Jessen
Tom Rymes wrote: > I dunno, I guess I'm not your mother, but then again, it seemed > pretty rude considering the guy offered the program for free and you > were criticizing the fact that he didn't develop a free linux app for > you, too. Not specifically directed at Toms reply - Gee, all Stephe