similar to: Incoming calls grind to a halt

Hello, So, I just got through a very stressful couple of hours. I''ve been running a rails application for a few months on a dedicated server. The application isn''t really that intensive and has been serving roughly 50k requests / day. It''s been handling really well, really fast, and all is good. The deployment configuration I am using is apache 2.2.3 with

Hi Devs and others, First, many thanks for 14+ years of personal usage rsync. I'm trying to rsync to an sshfs mount point instead of using ssh transport. The reason for this is that I'm using encfs to create files which land on the (untrusted) destination encrypted. After witnessing stalled rsync progress, a barely utilized internet connection and and idle CPU, I stripped this down

Hi. A friend of mine was doing real-time video decoding on Fedora Core 13 and he had a performance glitch (1/2 a second freeze) every 5-10 seconds. "top" showed flush-253:0 process at the moment of the freeze. Major device number 253 corresponds to device-mapper. I advised my friend to re-install his FC13 without LVM, to see if the glitch is related to LVM. After re-installing FC13

I am trying to use dovecot. I've got a GFS2 shared volume on two servers with dovecot running on both. On one server at a time, it works. The test I am trying is to attach two mail programs (MUA) via IMAPS (Thunderbird and Evolution as it happens). I've attached one mail program to each IMAPS server. I am trying to move emails around in one program (from folder to folder), and then

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

Hardware: Digium TE110P REV.C and REV.D Gigabyte GA-965G-DS3 Bios F8b cat /proc/cpuinfo .... model name : Intel(R) Core(TM)2 CPU 6600 @ 2.40GHz stepping : 6 cpu MHz : 2400.080 cache size : 4096 KB ... latest libpri, dahdi, asterisk as of tonight. linux: debian lenny After moving hardware around all slots, disabling all unused hardware with no

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

Hi guys I just registered an incoming number with iconnecthere and I'm trying to set up incoming calls from icconnecthere on my asterisk server. I took a look at john todds sample sip.conf and extensions.conf file but for some reason my incoming is still not working. At this point I wish to use iconnecthere merely for inbound calls. Also my asterisk server is behind nat. The following

so here it is, the problem that's been nagging me for the past 2 days: connected a box to my telco's NTBA <-> zap/asterisk. which works: box:/etc/asterisk# cat /proc/zaptel/1 Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [TE] layer 1 ACTIVATED (F7)" HDB3/CCS 1 ZTHFC1/0/1 Clear (In use) 2 ZTHFC1/0/2 Clear (In use) 3 ZTHFC1/0/3 HDLCFCS (In

Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio

Hi, I'm running CVS-HEAD-03/19/05-11:15:15 on Fedora Core 3 with Digium TE410P card. Calling into meeting rooms that have been configured with the p option works fine. From ZAP extensions the # key does not work to exit, however from SIP extensions the # key works fine. This makes me believe that somehow the DTMF doesn't get through the ZAP interface. After furter experimenting

Hi all, Our Asterisk box sends calls outbound via either SIP (through our VoIP provider) or an E1 PRI (directly connected via a TE410P). When we dial a number that is engaged via our VoIP provider we get the following on the Asterisk console (numbers and IP addresses changed to protect the innocent): -- Called 12345678@sip-outbound -- Got SIP response 486 "Busy here" back from

Dear All, I'm running SAMBA 2.0.5a on a Sun 450 running Solaris 7. In the past I've had an occasional problem with processes running amok and causing the server to grind to a halt, which I've put down to network glitches as they seemed to coincide with problems on the network. But today I've had 3 instances where the number of smbd processes has just steadily increased until

Hi there, I'm building a mail system for an ISP. We want to be able to turn on or off IMAP or POP3 access for certain customers- basically we want to be able to offer a customer POP3 access, unless they pay us some more money, in which case they get IMAP The username and password will remain the same, and are driven out of a postgres database. The best way I've though of doing this

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

Hi there, I'm using postgres as a userdb for dovecot. I'm am trying to get the quota plugin to work. I have a column in my table that returns the quota in KB. I am using the postgres expression "('maildir:storage=' || mailstore_control.mailbox.quota) AS quota" as recommended in the manuals. This expression is NULL to start with, which I take to mean unlimited