similar to: Record and then Read does not found file

Boa Tarde, Estou com problemas para autentica??o de meus computadores no dominio samba. Quando tento pelo pr?prio windows ou na m?o pelo console adicionar uma m?quina ao dominio, ela ? cadastrada com sucesso, ela fica em ou=Computers,dc=xxx,dc=xxx por?m o smbd a procura em people o que resulta que meus usu?rios n?o conseguem logar... assim ... ao inves de o sambra procurar em ou=Computers ele

libswfdec-gtk/swfdec_gtk_loader.c | 10 ++++++++-- libswfdec-gtk/swfdec_playback_alsa.c | 3 +-- libswfdec-gtk/swfdec_source.c | 32 ++++++++++++++++++++++++-------- libswfdec/swfdec_cached.c | 3 +-- libswfdec/swfdec_font.c | 19 ++++++++++++++----- libswfdec/swfdec_morphshape.c | 10 ++++++++-- libswfdec/swfdec_net_stream.c | 3 ++-

Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/ 003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should

Hi, I have a customer with a definitely low-tech need: he has a noisy storeroom where he wants to hear the phones ringing so he can leave the storeroom and pick up the phone in his office. So all I need is a loud SIP ringer. Does this even exist? I know paging amplifiers exist, but that`s not what I need. Mike -------------- next part -------------- An HTML attachment was

Hi, I have just followed the http://wiki.centos.org/HowTos/Nagios intruction and found some minor errors that I would like to correct. I would also like to add a section with NagViz to the Nagios HowTo. My user is matsk. Kind regards Mats -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am learning to write LLVM modules. I am using IRBuilder to insert calls to an existing function. The call is to be inserted at a random location in every function in the code. The call does get inserted, however, the module crashes later (after doFinalization()). The crash happens only sometimes, not always. These are the kind of instructions that I see in the generated code. %callins =

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards

part of extensions.conf: exten => 11,1,Answer() exten => 11,n,NoOp(CallerID : ${CALLERID(all)}) exten => 11,n,Playback(/tmp/welkom-tcs.alaw) exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1) ; wordt doorgerouteerd naar context open, maar indien gesloten : exten => 11,n,NoOp(Oproep tijdens winkel gesloten) exten => 11,n,Playback(/tmp/winkel-gesloten.alaw) exten =>

Hello, I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to CVS-HEAD, and realtime. Compiled no problem and now running, with realtime extensions and sip users in postgres (ODBC connection) database, trunking also works. I have looked on google, wiki, and this mailing list, along with talking to some peers, but to no avail. My problem revolves around voicemail. I have looked

Nice! Let me try to answer some questions, Before that I have to mention this is ORC version 2 APIs and this is where the project is moving forward. JITDylib is the symbol table, basically for a JIT Symbol it have an associated materializers, (you can think of it like an entity that generate the address for that symbol), Example: compiler are materializers. So to add symbols to your own JIT you

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

Hi List. Don't know if I already posted about this problem but, if I have I apologize for the double post. I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say "Good Morning" but, when I run the test I get the following error message saying that the file doesn't exist and it does: Night..............

Hi Guys, Just new to Asterisk and am completely stumped. I have created two accounts as instructed. Please see below for the config of the user accounts. [Peter] type=friend host=IP address disallow=all allow=ulaw allow=alaw callerid=Peter <6004> secret=XXXXXXX context=default port=9060 nat=force_rport,comedia deny=0.0.0.0

Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when

Hello, I am trying to create a simple IVR for testing.. What i did is to create a voice file from asterisk-gui. And i saw it created that under /var/lib/asterisk/sounds/record/ as deneme.gsm Then i tried to make a IVR menu and play that file. I tried exten=s,4,Playback(/record/deneme.gsm) exten=s,4,Playback(record/deneme.gsm) exten=s,4,Playback(deneme.gsm)

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hello list, it seems that a sound file is not present on my system, although I have made a standard install... [Sep 22 12:28:51] WARNING[22117]: file.c:650 ast_openstream_full: File vm-INBOXs does not exist in any format [Sep 22 12:28:51] WARNING[22117]: file.c:953 ast_streamfile: Unable to open vm-INBOXs (format 0x8 (alaw)): No such file or directory I do not find this particular soundfile

Dear useRs, I reran an analysis with bam (mgcv, version 1.7-17) originally conducted using an older version of bam (mgcv, version 1.7-11) and this resulted in the same estimates, but much lower standard errors (in some cases 20 times as low) and lower p-values. This obviously results in a larger set of significant predictors. Is this result expected given the improvements in the new version? Or

1.6.2: -- Executing [s at incoming-pstn-line:4] VoiceMail("DAHDI/4-1", "100 at default,u") in new stack -- <DAHDI/4-1> Playing '/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en') [Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File vm-intro does not exist in any format [Mar 22 17:15:46] WARNING[31145]:

AsmPrinter currently handles emitting the constants only in the default data space. The attached patch enhances those APIs to take address space attribute and allow emitting of constants in the specified address space. Please let me know your comments on the same. - Sanjiv -------------- next part -------------- An HTML attachment was scrubbed... URL: