similar to: Weird behaviour

Red Hat Enterprise Linux Server release 6.5 (Santiago) selinux is disabled. The following commands were all run on the RedHat Server on which I am running samba. *The following ports are open* 5 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 state NEW tcp dpt:137 6 ACCEPT tcp -- 0.0.0.0/0 0.0.0.0/0 state NEW tcp dpt:138 7 ACCEPT tcp --

Hello, I am trying to migrate emails from: Mountain Lion OSX 10.8.5 (dovecot: 2.0.19apple1) to: RHEL 7.0 (dovecot: 2.2.10) Using command: dsync -m /Library/Server/Mail/Data/mail/XYZWXYZW-XYZW-XYZW-XYW-XYZXYZXYZXYZ/ -u giedriust mirror giedriust at 192.168.xx.xx root at 192.168.xx.xx's password: dsync-remote(giedriust at domainname.com): Error: dsync(local): Remote dsync doesn't use

Hi, Running samba sernet 4.1.6-7, I've noticed the load avg slowly / steadily creeping up (.e.g > 100). I'm now noticing that several smbd processes are at 100%. I don't actually notice that much bandwidth usage on the system (e.g. iptraf/iftop). Any idea what's causing this? Restarting smbd helps for a few days, but then the high load avg returns. Thanks, Sabuj

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, I've callcenter and our queue members want to see on their IP phone's display queue's name , from which incoming call was originated, for example "<client's_number> -> Sales". This problem appears when one member can belong to couple queues. Work around would be setting calling name with such information. Maybe there is another way (setting SIP

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

Hello, I've this situation: 300+ simultaneous calls and dialplan like this: exten => _X.,1,Answer() exten => _X.,2,DEADAGI(check_status.php) exten => _X.,3,Dial(SIP/other/${NUMBER}) exten => _X.,4,Hangup exten => h,1,DEADAGI(cdr.php) When project is running , I had a lot of defunct php scripts (I've exceed mysql connection limits and so on, deadagi help a bit). The

Hi, I have problems with asterisk and hylafax+ iaxmodem. I can successfully send faxes to Panasonic KX-FT932 fax, but with Xerox WorkCentre M20i I have problems: No carrier. This is hylafax log, maybe you can suggest me where to find ... Oct 17 07:38:48.22: [22428]: SESSION BEGIN 000000041 180037052390906 Oct 17 07:38:48.22: [22428]: HylaFAX (tm) Version 4.4.2 Oct 17 07:38:48.22: [22428]: SEND

Hello, I've asterisk 1.4.22. I need to that the first conference user hears "You're the only conference user..." . When the second user joins (without recording his name) , the first user only hears "new user have join" , when the third user joins to conference, others hear "new user have join" and so on. I'll try to do this with meetme, but it always

Hello, I have created a dynamic conference into two languages (english and russian). Client calls to confrence number and interactive choose the language. Meetme runs with 'dMi' options. Everything works perfect if one conference room clients have choosed the same language. If clients had choosed different language , there is a problem with user join/leave announcements. For example:

Hello, Is it possible, that during the call one side , for examples clicks the button on the web, and this call starts recording? It's possible with asterisk feature automon and DTMF. So it is possible to start recording the channel using AMI or ... ? Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

hello, I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read application accepts # sign, So is it possible? And maybe there is a workaround? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/2f2b443c/attachment.htm

Hello, I want that after client and queue member call would be established, cmd queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This is my example of ael : context QUEUE { _X. => { Ringing(); Wait(4); Answer(); Queue(${Queue},wr,,,60,,,check-record); Hangup(); }; }; macro check-record() {

Hi, could someone help me with ssl certificates? i have mycert.pfx file (client certificate) and CA certificate ca.cer. i far as i know, ruby doesn''t understand pfx format, so i''ve converted it to pem format. in viewer pem looks like: Bag attributes blabla Key Attributes blabla ---begin rsa private key--- blabla ---end rsa private key----- --begin certificate-------- blabla

Hi, I have problem with Asterisk 1.6.0.1. I need to change language for playing prompts in Lithuanian. But in Asterisk 1.6.0.1 version everytime plays in English, but in Asterisk 1.4.x I haven't any problem. Maybe it is a bug ...? So I paste my test dialpan and prompt's locations. I hope this helps you. Files are: [root at voip asterisk]# find /var/lib/asterisk/sounds/test -name

Hello, Is it possible to do like this: play a sound file (if needed play in loop) while php agi script finishes work ? And how to do this? When on my server is huge load , I don't want that client hears silent , but hears music. Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working with asterisk 1.6. And I have success using func_odbc with one row query results (SELECT source,destination from cc WHERE ... ): exten => s,1,Ringing exten => s,n,Wait(4) exten => s,n,Answer exten => s,n,Set(ARRAY(NUMBER,REALNUMBER1,REALNUMBER2,STATUSAS)=${ODBC_GETVARIABLES(${NUMERIS})}) exten => s,n,Verbose(1| ${NUMERIS}, ${REALNUMBER1} ${REALNUMBER1},

Hello, Maybe there is the easiest way to compile additional my module without recompiling all asterisk? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100107/cfe8f0b7/attachment.htm

Hello, I want to know what is timeout for MS SQL connection? My config is: [mydb] enabled => yes dsn => MYDB pooling => yes limit => 200 share_connections => no username => login password => password pre-connect => yes backslash_is_escape => no In the peak , I can see : ODBC DSN Settings ----------------- Name: mydb DSN: MYDB Pooled: Yes Limit: 200

Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/d29c02b8/attachment.htm