[asterisk-users] Brazilian.
Some brazilian here on list? ___ --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
RES: RES: [Asterisk-Users] CDR MySQL
Yes! I´m using latest version... I see: int retries = 5;. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviada em: quinta-feira, 6 de outubro de 2005 10:27 Para: asterisk-users@lists.digium.com Assunto: Re: RES: [Asterisk-Users] CDR MySQL There is a recent fix to cdr_addon_mysql.c that added 5 retries and a timeout variable to the conf file. You might want to try that. Brett On 10/6/2005, "Jozeph Brasil" <[EMAIL PROTECTED]> wrote: >I don´t have access to MySQL administration! :( >I´m using a remote database to store... > >-Mensagem original- >De: Steve Davies [mailto:[EMAIL PROTECTED] >Enviada em: quinta-feira, 6 de outubro de 2005 09:48 >Para: Jozeph Brasil >Assunto: Re: [Asterisk-Users] CDR MySQL > >The usual "fix" is to reconfigure MySQL - It is the database that is >causing the timeouts. I do not know how this is done. Perhaps Google >for "MySQL connection timeout" and go from there? > >Regards, >Steve > >On 10/6/05, Jozeph Brasil <[EMAIL PROTECTED]> wrote: >> Hi Steve, >> >> Thank you for reply! :) Do you know if anyone have been patched to solve >this? >> >> Quoting Steve Davies <[EMAIL PROTECTED]>: >> >> > I do not know for sure, but in my experience, the most common cause of >> > this error is an application which assumes a database connection will >> > stay open forever. >> > >> > Some MySQL installs have a timeout on idle connections. It should be >> > possible to set this to a very-long or infinite timeout. >> > >> > Hope that helps, >> > Steve >> > > > >___ >--Bandwidth and Colocation sponsored by Easynews.com -- > >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] CDR MySQL
I don´t have access to MySQL administration! :( I´m using a remote database to store... -Mensagem original- De: Steve Davies [mailto:[EMAIL PROTECTED] Enviada em: quinta-feira, 6 de outubro de 2005 09:48 Para: Jozeph Brasil Assunto: Re: [Asterisk-Users] CDR MySQL The usual "fix" is to reconfigure MySQL - It is the database that is causing the timeouts. I do not know how this is done. Perhaps Google for "MySQL connection timeout" and go from there? Regards, Steve On 10/6/05, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi Steve, > > Thank you for reply! :) Do you know if anyone have been patched to solve this? > > Quoting Steve Davies <[EMAIL PROTECTED]>: > > > I do not know for sure, but in my experience, the most common cause of > > this error is an application which assumes a database connection will > > stay open forever. > > > > Some MySQL installs have a timeout on idle connections. It should be > > possible to set this to a very-long or infinite timeout. > > > > Hope that helps, > > Steve > > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR MySQL
Hi Steve, Thank you for reply! :) Do you know if anyone have been patched to solve this? Quoting Steve Davies <[EMAIL PROTECTED]>: I do not know for sure, but in my experience, the most common cause of this error is an application which assumes a database connection will stay open forever. Some MySQL installs have a timeout on idle connections. It should be possible to set this to a very-long or infinite timeout. Hope that helps, Steve On 10/5/05, Jozeph Brasil <[EMAIL PROTECTED]> wrote: Hi Asteriskers! I've enable CDR to store data on a remote machine using MySQL. But I have a problem. Analyzing the log, I see some ERROR messages as: -- SIP/21-3787 is ringing == Spawn extension (default, 21, 1) exited non-zero on 'SIP/21-ce14' Oct 5 13:22:54 ERROR[8576]: cdr_addon_mysql.c:161 mysql_log: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query This occurs every time that extension hangs up the call. Anyone know why asterisk lost connection during query? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR MySQL
Hi Asteriskers! I've enable CDR to store data on a remote machine using MySQL. But I have a problem. Analyzing the log, I see some ERROR messages as: -- SIP/21-3787 is ringing == Spawn extension (default, 21, 1) exited non-zero on 'SIP/21-ce14' Oct 5 13:22:54 ERROR[8576]: cdr_addon_mysql.c:161 mysql_log: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query This occurs every time that extension hangs up the call. Anyone know why asterisk lost connection during query? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR MySQL
Hi Asteriskers! I’ve enable CDR to store data on a remote machine using MySQL. But I have a problem. Analyzing the log, I see some ERROR messages as: -- SIP/21-3787 is ringing == Spawn extension (default, 21, 1) exited non-zero on 'SIP/21-ce14' Oct 5 13:22:54 ERROR[8576]: cdr_addon_mysql.c:161 mysql_log: cdr_mysql: Unknown connection error: (2013) Lost connection to MySQL server during query This occurs every time that extension hangs up the call. Anyone know why asterisk lost connection during query? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording calls
Hi guys, I need to record all incoming calls. Anyone know how to do this? Thanks, Jozeph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP dialing in two extensions
Hi guys, Is it possible to make Dial to call two extensions at the same time? I want when the user pressed extension it call to two SIP phones at the same time... Who wakeup first get the call... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] unavail and busy.
Hmmm
Thank you Eric.
Do you know where can I see all ${DIALSTATUS} available?
-Mensagem original-
De: Eric Wieling [mailto:[EMAIL PROTECTED]
Enviada em: segunda-feira, 13 de setembro de 2004 11:22
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Assunto: Re: [Asterisk-Users] unavail and busy.
On Mon, 2004-09-13 at 09:02, Jozeph Brasil wrote:
> Hi guys,
>
>
>
> What is different and the "context" to play unavail message and busy
> message?
>
> When a SIP connection is unregistered, voicemail will play unavail
> message, right?
No. If there's a n+101 Asterisk will jump to that if the destination is
busy, unavailable, lagged, or not registered. If you want to know the
ACTUAL cause of the the call failing you need to look at ${DIALSTATUS}
and figgle with dialplan logic.
--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
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[Asterisk-Users] unavail and busy.
Hi guys, What is different and the “context” to play unavail message and busy message? When a SIP connection is unregistered, voicemail will play unavail message, right? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail
Hi everybody, What´s necessary to do the voicemail work fine with Brazilian Portuguese? Look my log files: -- Playing 'vm-login' (language 'pt') -- Playing 'vm-password' (language 'pt') -- Playing 'vm-youhave' (language 'pt') -- Playing 'digits/1F' (language 'pt') -- Playing 'vm-message' (language 'pt') -- Playing 'vm-Olds' (language 'pt') --> The correct is only OLD, Olds is only when more then 1 message. -- Playing 'vm-onefor' (language 'pt') -- Playing 'vm-messages' (language 'pt') -- Playing 'vm-first' (language 'pt') -- Playing 'vm-message' (language 'pt') == Parsing '/var/spool/asterisk/voicemail/default/1234/Old/msg.txt': Found -- Playing 'vm-received' (language 'pt') -- Playing 'digits/day-2' (language 'pt') -- Playing 'digits/pt-e' (language 'pt') -> There isnt necessary. -- Playing 'digits/20' (language 'pt') -- Playing 'digits/4' (language 'pt') -- Playing 'digits/pt-de' (language 'pt') -- Playing 'digits/mon-7' (language 'pt') -- Playing 'digits/pt-de' (language 'pt') -- Playing 'digits/2' (language 'pt') -- Playing 'digits/pt-e' (language 'pt') --> this need to jump to after 1000. -- Playing 'digits/1000' (language 'pt') -- Playing 'digits/4' (language 'pt') -- Playing 'digits/at' (language 'pt') Asterisk try to play -23 sound file? I don´t understand this error... Sep 11 10:12:18 WARNING[294926]: file.c:475 ast_openstream: File digits/-23 does not exist in any format Sep 11 10:12:18 WARNING[294926]: file.c:779 ast_streamfile: Unable to open digits/-23 (format ILBC): No such file or directory -- Playing 'digits/20' (language 'pt') -- Playing 'digits/3' (language 'pt') --- Need to play HOUR.gsm (pt-horas) -- Playing 'digits/pt-e' (language 'pt') -- Playing 'digits/13' (language 'pt') --- Need to play MINUTES.gsm (pt-minutos) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
Hmmm, Flash work for IAX? -Mensagem original- De: Eric Bart [mailto:[EMAIL PROTECTED] Enviada em: sábado, 31 de julho de 2004 02:26 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! Flash don't work for sip - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 4:17 PM Subject: RES: [Asterisk-Users] Softphone - Freeware?! > I have one X100P installed with two SIP extensions using X-Lite, I just > would like to transfer the call to another SIP extension; Just a > "Flash"+"Extension"+"Hangup CALL"... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
Hi, I can´t connect to your website... :( Is it offline? -Mensagem original- De: Dan [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 11:12 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! Hi, Have you tried DIAX? It is an IAX2 based soft phone and it is free. Check: http://www.laser.com/dante If you need help, don't hesitate to send me a mail. Best regards, Dan P.S. Use the address from this mail instead of [EMAIL PROTECTED] - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 4:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! Hi everybody, What is the most complete Softphone application freeware? X-Lite is very CooL, but the free version don´t support transfers... :( Anyone know, a windows softphone free application that I can use all Askterisk Resources? Congratulations, Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a "Flash"+"Extension"+"Hangup CALL"... Thanks for all! -Mensagem original- De: Eric Bart [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 10:51 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! axra will do. it's an add-on that will give consultative transfer to X-Lite (and others). see below : --- New application for asterisk : axra axra runs separately. developped in C++. it dialogs with asterisk through agi calls and through the manager api. it proccesses phone calls through the dial plan (agi) and concurently through the manager api. axra currently provides consultative transfer for SIP and IAX2 phones. this should easily be extended to any phone technology. hopefully, axra will soon provide 3 way calling. there are two tranfer functions : PreTransfer and CTransfer each should be implemented in the dial plan like : exten => 76,1,AGI(axraagi|PreTransfer) exten => 76,2,Hangup exten => 77,1,AGI(axraagi|CTransfer|auto) exten => 77,2,Hangup you may choose other extensions than 76 & 77. you may omit 'auto' when a call is transfered to PreTransfer (76), the call is parked and waits for a transfer. if the timeout occurs, the call is ringed back. if you call PreTransfer (76) directly, the parked call (if any) is immediatly ringed back. when a call is transfered to CTransfer (77), the call is linked to the pretransfered (parked) call. if no pretransfer exists the call is pretransfered just like 76 was dialed. however, if 'auto' was specified, axra will try to link the call to the oldest live channel attached to transferer's phone. http://www.byortek.com/asterisk/axra-2004-07-29.tgz Please download, read REAME and INSTALL. Any feedback greatly appreciated. - Original Message - From: "Jozeph Brasil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, July 30, 2004 3:30 PM Subject: [Asterisk-Users] Softphone - Freeware?! > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
Hello, I try to google it without success... :( Sua pesquisa - "enable transfers" site:www.voip-info.org - não encontrou nenhum documento correspondente. Sugestões: - Certifique-se de que todas as palavras estejam escritas corretamente. - Tente palavras-chave diferentes. - Tente palavras-chave mais genéricas. - Tente usar menos palavras-chave. No match! :( -Mensagem original- De: Brian McSpadden [mailto:[EMAIL PROTECTED] Enviada em: sexta-feira, 30 de julho de 2004 10:45 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! We have found X-Lite to be one of the better softphones for Windows. It doesn't support SIP transfers, but it will do the # transfer from within *. Just hit pound during a call and you'll hear Allison say "Transfer", punch in the number you'd like to transfer to and away you go. You do need to have this enabled in the dialplan dial strings to enable transfers. See the Wiki for some help with this. http://www.voip-info.org On Fri, 30 Jul 2004 10:30:12 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi everybody, > > What is the most complete Softphone application freeware? X-Lite is > very CooL, but the free version don´t support transfers... :( > Anyone know, a windows softphone free application that I can use all > Askterisk Resources? > > Congratulations, > Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Softphone - Freeware?!
Hi everybody, What is the most complete Softphone application freeware? X-Lite is very CooL, but the free version don´t support transfers... :( Anyone know, a windows softphone free application that I can use all Askterisk Resources? Congratulations, Jozeph Brasil. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Please, Give-me more informations about your comercial web front end for *. -Mensagem original- De: Paulo H. Mannheimer [mailto:[EMAIL PROTECTED] Enviada em: quinta-feira, 29 de julho de 2004 08:15 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER * Hi, we are interested. We have developed a comercial web front end for * administrations (mailboxes, voicemail, platform status), as well as a visual tool for dialplan development. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING message - ALSA
Title: Trouble compiling asterisk-addons MySQL
Hello all,
I receive a warning message every time I start my
Asterisk.
Rate not correct, requested 8000, got 7999
I see in the source and FOUND where is
gerating this warning message.
if (rate != DESIRED_RATE) { ast_log(LOG_WARNING, "Rate not correct, requested %d, got %d\n", DESIRED_RATE, rate); }
Anyone know HOW TO FIX
this problem?
[Asterisk-Users] Confused.
Hi all,
I´m using X-Lite as
SoftPhone in Asterisk. I have configured this:
[101]
;Turn off silence
suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends
NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=jozeph
callerid="Jozeph
Brasil" <5678>
host=dynamic
nat=yes
; X-Lite is behind a NAT router
canreinvite=no
; Typically set to NO if behind NAT
;disallow=all
allow=gsm
; GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
When I try to connect
X-Lite from another network follow this model:
WORK (192.168.1.X) ß à FW-NAT ß à INTERNET ß à ASTERISK SERVER
That machine connect OK!
But, when I try to
connect using this model:
WORK (10.0.0.X) ß à 10.0.0.254 (ASTERISK SERVER)
I can´t connect to the
server why that?
Asterisk Server have 2
network cards... internal and external internet card.
My SIP.CONF are listen
0.0.0.0.
[Asterisk-Users] [br] ---> indications.conf
Hello all, Me again! How to use [Br] on indications.conf file? When I set loadzone = br; defaultzone = br; on /etc/zaptel.conf I receive an error... Maybe I need to setup it from other file... Anyone can help? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Play CD!
I do that. But when I play MusicOnHold the music is played slowly! I don´t know why... but is how the bitrate is playing with a different number. -Mensagem original- De: Chris Foster [mailto:[EMAIL PROTECTED] Enviada em: sábado, 24 de julho de 2004 01:37 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Play CD! On Fri, 23 Jul 2004 23:29:19 -0300, Jozeph Brasil <[EMAIL PROTECTED]> wrote: > Hi all, > > Is it possible to play a CD has MusicOnHold? > > Thanks, > Jozeph > Why don't you just rip the CD to MP3? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Documentation
Hello all, Anyone know where can I get a complete source that describe "all" options available in the configuration files? I like to know all available options in configuration files with a description and a correct syntax. Another think I would like to understand is what´s the real function of all files in the modules directory... I found someone in voip-info.org, but don´t have "all" files described. Thank you for any help! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Play CD!
Hi all, Is it possible to play a CD has MusicOnHold? Thanks, Jozeph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie - card X100P.
Hello all, I´m starting with Asterisk and will do frequently question. Before ask anything, I have consulted voip-info.org and some documents of asterisk. When I start Asterisk, I receive this error: Jul 21 22:05:30 WARNING[196620]: chan_oss.c:238 sound_thread: Read error on sound device: Resource temporarily unavailable If I try to use alsa, using noload on oss, I receive: Jul 21 22:18:42 ERROR[16384]: chan_alsa.c:297 alsa_card_init: snd_pcm_open failed: No such file or directory Jul 21 22:18:42 ERROR[16384]: chan_alsa.c:297 alsa_card_init: snd_pcm_open failed: No such file or directory Jul 21 22:18:42 ERROR[16384]: chan_alsa.c:432 soundcard_init: Problem opening alsa I/O devices My sound card is correct configured because when I try to play a file using: play somefile.gsm I listen it. My other error is when I try to make a call: Jul 21 22:05:57 NOTICE[229390]: rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received Jul 21 22:05:58 NOTICE[229390]: rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received If anyone can help I will be very great. Please, if you have a X100P, give-me your config directory compressed that can I use it as example to configure my Asterisk. Please! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
