similar to: Implement missing features in Meetme application

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

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..............

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

Hi gang, I have a "fun" one for you. I'm not getting the quality of sound I want out of GSM, so I'm trying to make my files into .WAV (.wav) format. Here is the "file" output for 5 files: file *.WAV cents.WAV: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz dollars.WAV: RIFF (little-endian) data, WAVE audio, Microsoft

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Hello, Talking with Soren Sratje about Caller ID in Brazil, we compare ours DTMF tones captured by ztmonitor. wcfxo correctly recognize the "DTMF CLIP" and asterisk shot the AST_STATE_PRERING correctly. But the DTMF tones are not reconized. In the chan_zap.c, the code: if (f->frametype == AST_FRAME_DTMF) { (...) Does not occurs because the frametype is always reconized as voice

Hi all, I'm trying to set up a very simple shared mail server, where each 'domain' is a system user, i.e. 'example.com' is a real user with /home/example.com/, a uid of 5001 (and gid 'example.com' of 5001). Each domain\user has their own maildir inside their home, and a plain passwd-file with the virtual mail users associated with that domain. Version & config

Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx at Accueil_HNO:2] BackGround("SIP/SIP000005-0000000c", "Fermeture") in new stack [Jun 16 07:35:06] WARNING[7634][C-00000006]: file.c:701 ast_openstream_full: File Fermeture does

Hi R users: I don?t know if anybody have had the same problem with 'step()' funtion. If I type the commands ------------------------------------------------------------------------------- testData<-read.table("testData.dat",header=T) model1J<-glm(MCHNV~offset(Offset1),data=testData,family="poisson") step(model1J,direction="forward",

Hi, I cannot get two Polycom 600 phones to bridge natively. My sip.conf has canreinvite=yes for both phones. They connect, and I can talk as usual, but sniffing shows the RTP stream is routed through Asterisk. The exact spot where the attempt to natively bridge fails is in rtp.c, line 1281 (CVS from October 8, 2003): f = ast_read(who); if (!f || ((f->frametype == AST_FRAME_DTMF)

I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same

Hi all, I'm trying to set up a very simple shared mail server, where each 'domain' is a system user, i.e. 'example.com' is a real user with /home/example.com/, a uid of 5001 (and gid 'example.com' of 5001). Each domain\user has their own maildir inside their home, and a plain passwd-file with the virtual mail users associated with that domain. Version & config

(alternative title - what did I do wrong? or suggestions to make this work) Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb /usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48: error: ? does not name a type ) 1.6 did compile and almost works. 'cept it thinks the .gsm files are not played. from

Hello list, I have a file to be played in wav-format. I thought Asterisk would automatically take the wav-file and translate it to the codec used, but I see this : [Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File /var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any format [Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to open

Buenos días, Para una media ponderada con segmentación prueba a usar compmeans() del paquete descr. Un saludo, Manel. 2011/7/29 <r-help-es-request@r-project.org> > Envíe los mensajes para la lista R-help-es a > r-help-es@r-project.org > > Para subscribirse o anular su subscripción a través de la WEB > https://stat.ethz.ch/mailman/listinfo/r-help-es > >

Hola a todos. Ya he escrito sobre esto pero vuelvo a no ser capaz de resolver una cosa Tengo este archivo año   mes      fecha  puerto           barco   origen       arte            2007 enero 2007-01-04 Ribeira  juan        Norte     PAREJA      2007 enero 2007-01-04 Ribeira    luis        -Norte     BETA    2007 enero 2007-01-04 Ribeira  pedro     Norte     PAREJA      2007 enero

msz De: Fern?ndez-L?pez, Ambito Iberoamericano, Paseo de la Castellana 223, Madrid. [1]InEnglish - [2]EnEspanol Caros amigos luso-brasileiros, ? de se perguntar se as "patrulhas ideol?gicas" esquerdistas estar?o impedindo que os ?ltimos artigos do ex preso pol?tico e escritor cubano Armando Valladares - que abordam delicados aspetos das rela??es entre o regime

I've recently been working with some California county-level data. The counties can be referred to as either FIPS codes, eg F060102, friendly names such as "Del Norte County", names without 'County' on the end, names with 'CA' on the end ("Del Norte County, CA"). Different data sets use slightly different forms and putting them all together is a pain. So I

Hello, every time I browse the printer folder in any of the 2 samba servers from a W2K Wks, I get this error: [2002/06/02 15:58:50, 0] smbd/service.c:make_connection(252) contmaia334 (10.240.5.116) couldn't find service ::{2227a280-3aea-1069-a2de-08 002b30309d} I have a Samba Server as PDC and another that gets the authentication from the PDC (=domain). This error appears in both two.