[Asterisk-Users] How to change the packet size

Hi, We observed the packet size used in asterisk is about 20 ms. We would like to know if is possible to change this value to 10 or 30 ms for example. If so, how could I change it? Thanks in advance and best regards __ Do You Yahoo!? Tired of spam?

[Asterisk-Users] Doubts about the Monitoring command

Hi all, I have some doubts concerning the way asterisk records calls using the Monitor command. I ´ve done some jitter and packet loss tests in a such way that, from asterisk 1, I send a file to asterisk 2 and record this file in asterisk 2 using the Monitor command. To simulate the jitter and pac

[Asterisk-Users] Portuguese (Brazil) configuration setup

Hi all, I´d like to know if it is possible, in asterisk, to modify the configuration setup from english language to portuguese (Brazil) one. If so, how can I modify it,ie which files do I need to modify to get this setup working? Thanks in advance and best regards Guild Jackson

[Asterisk-Users] Doubts regarding g726 - 16 bits setup

Hi all, I would like to make a call using the asterisk IAX with g726 - 16 bits codec. How could I configure it in the iax.conf file. Do I need to modify the file like this? . . disallow = all allow = g72616k . . I have tried it but it hasn´t worked. Thanks in advance and best regards Guild

[Asterisk-Users] How to decrease the speech volume for record?

. Can someone help me with that? Thanks in advance and best regards Guild Jackson __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list

[Asterisk-Users] 404 error found when making SIP point to point calls

Hi I would like to make point to point SIP calls between two digium with asterisk. I have got the asterisk working with the digium fxs interface but can´t get the SIP session working yet. I have configured the sip.conf and extensions.conf files the way that I can have calls between the extensions