similar to: codec bridging

hi all ! we've got IAX2 protocol working between several Asterisk servers. Now we are concerned with doing bandwidth management to maintain an acceptable voice quality. We thought of prioritizing the udp traffic. ( Giving a high priority to those IAX2 udp ports.) I know that IAX2 uses udp/4569. Is there any other traffic/ports that we need to consider for bandwidth shaping w.r.t IAX2.

Setup as follows: [private*] - Natting Router - [public*] [private*] cannot register via IAX2 correctly while [public*] is running. Status remains UNKNOWN even after minutes, calls from [public*] to [private*] are not possible. Console output of [public*]: | *CLI> iax2 show peers | Name/Username Host Mask Port Status | iaxtest/iaxtest (Unspecified) (D)

hi ! I have asterisk with Licensed G.729 codec enabled. Whenever I make a call using this codec a warning apears as, WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames WARNING[18450]: File dsp.c, Line 1106 (ast_dsp_process): Unable to detect process 256 frames

hi! I wanna do some arithmatic operations (addition and substraction -integer operation) inside extensions.conf. Is there a simple way to do this. If I do yy = ${xx} + 1 // say "xx" is initialized to '0' the resulting "yy" will show "0 + 1" Obiviously not the result I need. Any help !!!!! denzel.

Hi All, What the support that asterisk has to send/receive Faxes? Can I plug a FAX machine in the a FXS extension and send out Faxes? What's the codec I need to use? g.711? Also can we receive a FAX into a FXS extension in Asterisk PBX? Also I need to know if we can send/receive FSK data from/to an extension plugged into Asterisk PBX? For example if there's a phone model which can send

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

Hi, I have a function that generates a set of data but I am having problems determining the parameters using the nls fitting procedure. #### "MH"<-function(field,diameter,mu=10e-7,sig=0.1,Ms=100,chi=0){ #variables mu, sig, chi, Ms #input: field and diameter #all in CGS rho <- 5 kb <- 1.38e-16 t <- 300 length.d<-length(diameter) length.H<-length(field)

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec

Hello everyone, I am trying to use the following code: dat<-read.table(file="C:\\Documents.....txt") class2<-read.table(file="C:\\Documents.....txt") class1<-read.table(file="C:\\Documents .....txt") dat<-as.data.frame(dat) dimnames(dat)[[1]]<-paste("g",c(1:nrow(dat)),sep="")

hi! I've got cisco 7960G working with * box. Calls could be Blind Xfered through the phone but not the supervised transfer( Message on the phone: Transfer failed). Even when I put the caller on hold and resume it later, I can't hear the other side but the otherside can hear me. (It shows as the line is connected though. Yet the respective caller entry blinks.) Any suggestions most

hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated.

hi! I'm looking for a robust hardware IP phone which supports SIP protocol inorder to implement a call centre. Have anyone used CISCO SIP phones (eg:- 7960G ) with asterisk. From what I know these CISCO IP phones are very robust and feature rich. Yet I'm nervous whether * don't like CISCO at all. Thoughts are most welcome. denzel. -------------- next part -------------- An HTML

hi! I've got a asterisk system running with around 50 per calls per minute. I've connected * to internal pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have to live with or is it a bug in CVS code ? denzel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, all. I have problems with iax2, when try to communicate with one third server, asterisk reports the following errors in server's, could help me? Server A it speaks It with C in iax and Server B it speaks with D in iax, but Server A it does not obtain to speak with B in iax, reports the following error in server B "chan_iax2.c:5749 socket_read: Host 200.xxx.xxx.xxx failed you

Hello list, I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten. To simplify the problem I've brought it down to the following scenario: - 3 Asterisk Server A,B and C (central). - A and B both register with C. Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.

Gracias a todos por las recomendaciones. Ya me he puesto en contacto con el autor y le explicado donde esta el error. Es tan fácil como que han cambiado la estructura de los objetos tipos lmer desde la versión 2.15.3 con lo que su paquete dejo de funcionar. Por suerte la nueva estructura conserva por lo menos la parte necesaria para hacer sus funciones. Se localizar el fallo y como arreglarlo, lo

As of release samba-1.9.16p11, the X option (exclude file) for tar creates doesn't work recursively; if the exclude file is a directory, files that are in the directory or subdirectories are not excluded from the tar create. I would like the exclude feature to work recursively, and attached is a patch to do it. The exclude feature works the same way for both create and extracts. ***

Hace unos meses os escribir para comunicaros que había un fallo en esta función. Como os prometí os comento la respuesta por si alguno está interesado en utilizar el paquete LMERconvenientsfucntions Dear Javier, The package has been updated and should work for you fine now. Note that function mcp.fnc does not return the fourth plot (dffits) anymore. We still have to figure out a way to compute

Hello there, I wish to solve the following nonlinear System of equations: + u1 - Vmax11*S1/(S1 + Km11 *(1 + S2/Km21)) - Vmax12*S1/( S1 + Km12 *(1+S2/Km22)) == 0 + u2 - Vmax22*S2/(S2 + Km22 *(1 + S1/Km12)) - Vmax21*S2/( S2 + Km21 *(1+S1/Km11)) == 0 + Vmax11*S1/(S1 + Km11 *(1 + S2/Km21)) + Vmax12*S1/( S1 + Km12 *(1+S2/Km22)) - d1*P1 == 0 + Vmax22*S2/(S2 + Km22 *(1 + S1/Km12)) + Vmax21*S2/( S2 +