similar to: Mac Backup App

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

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Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

Hi Amit OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant

Hi, I've seen some chatter on NFS boards about kqueue being more reliable than inotify when used in NFSv3 and NFSv2. The chatter is a bit old so I don't know if it is true anymore. Anyone have pro/con experience with dovecot on the inotify/kqueue question when using NFS storage? I realize that kqueue is probably a bit slower and causes some delay with IDLE. Also, it may not really

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

I've been testing the debug version of AstTAPI, which worked for a few calls, then a bit later in the day (and ever since), when the call is hung up, the TAPI client doesn't get notified. Looking at the server logs, The TAPI message that is sent upon hangup, isn't being sent. exten => h,1,UserEvent(TAPI|TAPIEVENT: LINE_CALLSTATE LINECALLSTATE_IDLE) This is in the same context as

Hi, As far as I can see, the only way you can tell it's been a dry-run at the receiver-side, is to parse RSYNC_ARGs in the pre-xfer script. I think the only way is to crawl through the RSYNC_ARGs to find a value which: - starts with a single dash, and; - contains 'n', and; - comes prior to the "." value, signifying the start of the 'path args' (which seem inherently

Hi, We have a issue with a server pretty much just running dovecot, there's about 250gb of maildirs which has about 200 simultaneous connections to it running on freebsd 7. The machine has stayed up for about 8 days before it panics but sometimes panics after about 24 hours. We are running a PAE kernel (with ULE scheduler) as the machine has 2 (4 core processors) and 8gb of ram, dovecot is

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi All, I'm trying to debug DTMF issues I have with certain endpoint conferencing systems (external, 3rd party). On our A*k server I log DTMF, and I see that coming through in the log. What I'd like to see is what is sent onto our VoIP carrier over SIP. I can do a tcpdump of the packets, but what am I then looking for? Would it be in the RTP audio stream or within the SIP

Hi, I recently switched from courier imap to dovecot. With courier I had a working IDLE setup that informed me immediately when new mail arrived. With Dovecot it is different, sometimes i get an immediate result but most of the time it takes a rather long time for the notification to return to the client. For testing purposes I set mailbox_idle_check_interval = 1 and i now get the same

I'm getting this error message and I don't really understand what rsync is trying to tell me:- rsync: link_stat "/rdiffBackup/gradwell/Mail/." failed: No such file or directory (2) rsync error: some files could not be transferred (code 23) at main.c(977) [sender=2.6.9] Can anyone explain what it's saying please. /rdiffBackup/gradwell/Mail/ does exist and is

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Hi All, I'm looking for how to enable SIP Markers, or specifically, how to have the TIME reset when a call route changes. I'm debugging an issue, where a sip client we have switching to one-way-audio, when an asterisk server fruther down the call path dials out to the PSTN. Scenario is: SIP Client -> A*k1 -> A*k2 -> PSTN Provider/Gradwell -> O2 -> Mobile

Hi, the attached patch contains these changes: 2005-12-15 22:18 Vaclav Haisman <v.haisman@sh.cvut.cz> * src/lib/ioloop-kqueue.c: Fix IO_ERROR behaviour. * src/lib/ioloop-notify-kqueue.c: New file. * configure.in: Improve kqueue detection and handling. Cleanup. The configure.in changes that are not related to kqueue are there to make autoconf 2.59, automake 1.9.6 and libtool 1.5.20

I rewrote much of the kqueue code since the old code didn't handle properly the case when both input and output I/O handlers were added to the same file descriptor with different callbacks. This is done commonly in Dovecot, so I'm wondering why more people didn't complain about problems with it.. :) I don't anyway have access to any BSDs so the code is completely untested. Please

All, I've been experimenting with shortcodes for SIPgate etc. Passing calls to SIPbroker seems a good way to go, but the message I've had back from SIPgate is "we don't support SIPBroker"... So whats the easiest way to support SIP <> SIP network calling? At the moment, I've setup some local shortcodes (eg dial **777. to goto sipgate.co.uk) based on what Gradwell

Hi, I realise this is probably the wrong list for such a question, but I need a pointer to somewhere I can get some feedback on experience of (business class) voip providers for the UK? Situation is that we are currently with Gradwell and use them for an inbound/outbound single line for a business and their quality has gone from excellent to abysmal in the last few weeks. I'm sure they