search for: rvvvvvvvvvv

...gents via the legacy pbx (which is in sync with asterisk on span2)....This works perfectly fine until about 200 calls or so...After that time when asterisk tries to dial to the legacy pbx - the call drops with error "All are busy congested at this time" .the same is indicated on asterisk -rvvvvvvvvvv , but the spans are up and active at that time... can anyone throw some light on this ? >>> ZAPTEL.CONF span=1,0,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 >>> ZAPATA.CONF context=pri-pstn switchtype=euroisdn pr...

...(freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1 was answered. > Launching NoCDR() on Local/s at tc-maint-000002a4;1 [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 attempt_thread: Call completed to Local/s at tc-maint [2013-06-24 15:22:32] NOTICE[32678]: pbx_spool.c:402 att...

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i noticed the problem could be the module format_mp3.so not being loaded because not exists in my PC. in modules.conf i comment the line load => format_mp3.so and now it's works.This...

Does anyone else have DTMF issues with SIPPhone? When calling into my DID, and entering, say, 1002. Sometimes it will recognize it properly (rarely), other times it will receive something different. Such as, 1102 or 1000, etc. Has anyone else been having these issues? I'm only accepting ulaw and alaw, and my relevant sip.conf information follows: [sipphone] type=peer