Hello, on the recommandation of François Bergeret (F6HQZ), I tried to use app_rpt for a very specific application. This application is not ham-related so I'm sometimes lost in app_rpt's documentation as I don't understand everything. My problematic is very simple: on one side I have an audio connection initiated by SIP. On the other side I have a 4-wires analogic line (2 wires for rx and 2 wires for tx, that's the part which is common with the HAM world and that app_rpt is able to handle) conveying *only* the voice (there's no signalling at all). My goal is to bridge the two sides: as soon as the SIP connection is established, the audio should be forwarded on the "tx" line and the audio received on the "rx" line should be sent to the SIP side. And I try to use app_rpt to do this bridging. On the hardware side I'm using a Digium TDM400P with two FXO modules (since I don't want to feed current on the analogic lines). The zaptel configuration is: fxsks=3 # FXO RX pair fxsls=4 # FXO TX pair The Asterisk zapata configuration is: signalling=fxs_rx channel => 3 signalling=fxs_tx channel => 4 The app_rpt configuration is the smallest to I could come up with: [100] ; Node ID of first repeater rxchannel = Zap/3 ; Rx audio/signalling channel txchannel = Zap/4 ; Tx audio/signalling channel idrecording = test ; id recording [nodes] 100 = incoming@127.0.0.1,NONE Then in the Asterisk's extensions.conf, I create the application associated to my SIP id "testfxo": exten => testfxo,1,Answer() exten => testfxo,n,Rpt(100|D) exten => testfxo,n,Hangup() Using the "Dumb Control Phone" mode is the one which gave the best results (standard mode is refused, it accepts only IAX lines). In this configuration, I can connect via SIP to "testfxo" and I can hear my voice in the tx line and I can also hear (on the SIP side) the sound received on the rx line (but I have some echo).... this last a few dozen seconds at which point, there's a female voice which says "100" (the name of the repeater) and then I lost one direction of communication (analog->SIP). This side of the communication also happened to suffer from a delay of 1 second whereas the communication in the other direction seems without delay. For this setup to work, we had to polarize the analog lines by injecting some current. I don't know if this a requirement of the FXO module itself or if this just a signalling problem (ie app_rpt/Asterisk waits until the line seems to be connected before starting to really communicate). (If you know the answer, I'm interested, and if you know how to avoid this requirement I'm even more interested ;-)) After this lengthy explanation, let me ask some questions: - is there a way to configure app_rpt to just do the bridging without offering any control functions ? I bet that the delay I mentionned above is due to the operations made by app_rpt (DTMF decoding, etc). Is the non-Phone control mode supposed to be what I'm looking for? If yes, do you know why only IAX connections are accepted ? I tried to look in the code, but I didn't find any valid reason except that some authentication is made using IAX parameters. If this is the only reason, it should be easy to modify it to accept a SIP connection. - what's the difference in behaviour between before and after the announce of the repeater name ? - given my requirement I bet that I will create a new application based on app_rpt where I will strip most of the code and keep only the required parts. app_rpt is a big piece of code (>6000 lines) so if someone familiar with the code could point me to the most important functions to keep and play with, I'd be grateful. If you have seen big mistakes in the approach that I followed, feel free to correct me. :-) Thank you in advance for your help. And I also wish a quick recovery to Steve & Jim. -- Raphaël Hertzog -+- http://www.ouaza.com Freexian : des développeurs Debian au service des entreprises http://www.freexian.com