search for: unavilable

...2 inbound calls rang within a few secon= ds, asterisk would send the first to all phones, and then when tyring to se= nd the 2nd, would receive a BUSY message from the phones (because they were= busy processing a ring for the first caller), and the 2nd caller would win= d up going straight to the unavilable destination for the ring group, inste= ad of eventually ringing through to the phones after someone answered the f= irst call. I greatly appreciate your help & insight with this issue! - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ ....

Dear all, I noticed a very strange problem. When I tried using VoiceMailMain to record my unavailable message, the file does not get created even though I can find the corresponding mssage from asterisk: -- <SIP/2001-b6307d78> Playing 'beep' (language 'en') -- x=0, open writing: /var/spool/asterisk/voicemail/default/2000/unavail.tmp format: wav, 0x82828c8 --

hi My stations are behinds a firewall, the system is windows 2000 & 98, i use sjphone asterisk is on the internet gateway where is the firewall Shorewall the system is linux debian (sid) kernel 2.4.20 j do whaton http://www.automated.it/guidetoasterisk.htm (grateful Andy) to write my sip.conf but i can't call an external sip user. (an external user can call me) i try without asterisk with

Hi, I think these two patches are related to the topic: https://reviews.llvm.org/D41578 “[SCEV] Do not cache S -> V if S is not equivalent of V” * It’s committed. It can cause generation of redundant instructions. We’ve got regressions due to it. The biggest one is 7.31% regression in Spec2k6 401.bzip2 on Cortex-A57(AArch64). https://reviews.llvm.org/D42290 “[SCEV] Clear poison flags

I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten =>

Hi My head hurts... Can anyone help out here, my remote IAX can see my local IAX and visa versa, conversation starts, I can dial my remote (POTS) landline number, remote end answers, trys to route to local iax2, I see it start the conversation here, the extension (SIP) rings once and then it dies... Both ends are defined with accept IPADDRESS to keep it in the family and simple.. Debug info