Re: [asterisk-users] How to get the original SIP result code

You can use AMI Commands and run sip set debug from that you have to capture the response code. http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command Regards, On Thu, Aug 22, 2013 at 10:43 PM, Mordechay Kaganer wrote: > B.H. > > Hello, i'm using AMI Originate action (with asyn

[asterisk-users] How to get the original SIP result code

B.H. Hello, i'm using AMI Originate action (with async=true) to send outgoing calls to a SIP trunk (using asterisk-java library to connect to AMI). The problem is that in case of failed originate, OriginateResponse event is returning only the reason code which is sometimes not sufficient to deter

Re: [asterisk-users] is it possible to compile chan_h323 with 11.5.0?

ok, changed to ast_channel_writeformat ast_channel_readformat at least, got it compiled :-D 22.08.2013 13:24, Dmitry Melekhov пишет: Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing point

[asterisk-users] is it possible to compile chan_h323 with 11.5.0?

Hello! Tried to compile, but : [CC] chan_h323.c -> chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing pointer to incomplete type chan_h323.c:350: error: dereferencing pointer to incomplete type chan_h323.c: In function 'oh323_rtp_read': chan_h323