search for: unavailbl

Folks! I discovered some serious problem with several Sipuras 3000 but I don't know if the problem is with them or Asterisk. Basically, if I call a Sipura PSTN line, when there is a call already in progress, generally I get a 503 Sevice Unavailable, but if I try hard enough, I am able to get through and connect to dialed number. The other call gets disconnected but the originator of the

...9;m having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody seen this and if so how do I fix it? I've looked around on google and the list archives and it appears that there are others with similar problems with most people believing it to be a configuration problem. Since I don'...

...f the physical pipe. Typically, here in the office I have a burstable T1. The other office has a CIR of 762kbps or such. At the house I have ADSL with 128kbps uplink. At some number of calls, the channel will collapse. With a PRI, that's not a problem. Once all my T1 channels are busy, I get an unavailble, and I can inform the caller or take other action. With a IAX trunk, I have already observed (at the house) serious call/voice deterioration due to channel verload. How does one stop this? I.e. it would be very desirable to specify channel capacity (say xx number of simultaneous calls allowed) an...

...", &path) == 0 6433406 zfs_open() can leak memory on failure 6433408 namespace_reload() can leak memory on allocation failure 6433679 zpool_refresh_stats() has poor error semantics 6433680 changelist_gather() ignores libuutil errors 6433717 offline devices should not be marked persistently unavailble 6435779 6433679 broke zpool import 6436502 fsstat needs to support file systems greater than 2TB 6436514 zfs share on /var/mail needs to be run explicitly after system boots 6436524 importing a bogus pool config can panic system 6436526 delete_queue thread reporting drained when it may not be true...

Hello, I have my sip stuff seemingly working fine as well as my zaptel stuff working great... But I have a problem with sip registration timers (I'm guessing here). In my extensions.conf file I have this... exten => 2244,1,Dial,Zap/2|25 exten => 2244,2,Dial,Sip/brian|25 exten => 2244,3,VoiceMail,u2244 But if I close my sip phone and a call goes through it will still wait the 25

...exten => 1,*,Macro(checkmessage,${ARG1}) exten => 2,*,Hangup exten => s,3,VoiceMail2(s${ARG1}) exten => s,4,Hangup [macro-checkmessage] exten => s,1,VoiceMailMain2(${ARG1}) exten => s,2,Hangup If I do nothing during the greeting all behaves as I would expect (user is given the unavailble greeting, then a beep to leave the message) and the message is delivered properly. However, if I hit the * key during the greeting, I get the following error: == Spawn extension (macro-vmessage, s, 2) exited non-zero on 'SIP/2400-9055' in macro 'vmessage' Apr 14 17:43:06 WA...

...9;m having an odd problem with voice mail on a recent CVS of * where it appears not to detect a hangup on FXO and * will keep treating the call as new and continue leaving voicemails until the max has been reached. It will then continue trying to leave voice mails and basically makes the system unavailble to any further incoming or outgoing calls on that FXO..has anybody seen this and if so how do I fix it? I've looked around on google and the list archives and it appears that there are others with similar problems with most people believing it to be a configuration problem. Since I don'...

Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I

We fairly recently switched service providers for our 4 PRI circuits. Since that time, we started to notice some failed inbound calls. These calls terminate with an ISDN cause code 47 'resource unavailble'. Most of the time I see this error on the first or second channel on the second span in a trunk group (This is the providers trunk group for hunting, not an Asterisk trunk group). All the PRIs are setup as individual spans, we are not using NFAS. If the provider sets the hunt method to '...