Good morning. This is more of an FYI than a question for the group.. I was struggling with loss of call audio issues. The setup is an Asterisk 1.4 server with Linksys SPA942 phones, Bell PRI and Sangoma 101d card. Calls with a leg on the PRI would occasionally lose audio on the SIP end. So the person in the office stopped hearing the external person, yet the external person could hear them. Did not happen on internal (sip 2 sip) calls. Since it was occasional and seemingly random, I've been pulling my hair out for the last couple of weeks to find and kill it. I finally noticed a pattern.. this error in /var/log/asterisk/messages [Jul 2 09:13:46] WARNING[29627] rtp.c: Don't know how to represent 'f' I was having DTMF issues a while back and had enabled Sangoma's Hardware DTMF option (TDMV_HW_DTMF = YES). Disabling it last night and restarting the wanpipe interface seems to have killed the error and the audio loss problem.
I'm assuming dtmf 'f' has something to do with fax recognition. I'm also guessing the dtmf 'f' was fake, being accidentally manufactured by the Sangoma card, and triggered a stoppage in the flow of audio to the SIP extension from Asterisk. Meh. -Marc Marc Carrafiello Datex Billing Services Inc. --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
