search for: atif

...dovecot (version 2.2.9) with mdbox mail format. When I run dsync tool with "mirror" or "backup" parameters my source and destination directory synchronize correctly but if I delete some messages in user mailbox, deleted messages does not synced to destination. For example : atif at domain.com path is /mail/domain.com/atif/ and its size is 1GB. after first running the "dsync -u atif at domain.com backup mdbox:/backup/domain.com/atif/" command, size of "/backup/domain.com/atif/" is 1GB I deleted 300mb messages in atif mailbox then /mail/domain.com/ati...

Hi Atif make is a Unix's command which uses Makefile file for package's compilation. So after installing the complete development package from distribution disk, launch make. Ciao mauro

...$srcfile="/tmp/mycall"; $dstfile="/var/spool/asterisk/outgoing/mycall"; open(MYCALL,">$srcfile") || die "Cant't open file :$srcfile $!\n"; print MYCALL "Channel:IAX2/bali:bali\@nain/25\@atif\n"; print MYCALL "MaxRetries:2\n"; print MYCALL "RetryTime:60\n"; print MYCALL "WaitTime:30\n"; print MYCALL "Context:atif\n"; print MYCALL "Extension:22\n";...

on the asterisk site, it was stated while ago, how to download stable version. like cvs checkout -r v1-0_stable asterisk-addons zaptel libpri but now it's not their. is stable-version removed from the CVS ? or is their some different procedure ? thank you -- Atif

...d both KPhone and IaxComm for linux but receiving no audio from asterisk. sound is working fine, as I can listen playing files using PLAY or APLAY. KPhone is configured with DTMFmode=inband and codec is ulaw and IaxComm is configured with ilbc if somebody can sort out this Thank you regards, -- Atif

...AGI, otherwise it quits $AGI->get_data('ccs-getnumber','100000000','2'); print STDERR "dialing complete...\n"; **************************************************************************** Some one can sort out, where things are going wrong Thank you Atif 35,1 Top -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040507/a68b704d/attachment.htm

...erisk are not relayed back to the Calling Asterisk. Instead a 403 forbidden error message is sent back to the Calling Asterisk whatever the error response (503, 484, etc). Is there a way to relay back error responses through configuration scripts or do I have to dig in the source code -- Atif

...NVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:13@192.168.0.2> Proxy-Authenticate: Digest realm="asterisk", nonce="50b81cdd" Content-Length: 0 to 192.168.0.187:5060 Scheduling destruction of call '1e020TNnX5IvcvFu' in 15000 ms Found user 'chinee' Atif ________________________________________________________________ Sent via the WebMail system at convergence.com.pk

...XTEN:3},45,tT) 2 - exten => _01144808XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 3 - exten => _01144500XXXXXXX,1,Dial(${MAG}/${EXTEN:3},45,tT) 4 - exten => _011.,1,AGI(iax.agi) 4 - exten => _011.,2,Dial(${MAG}/${EXTEN:3},45,tT) 4 - exten => _011.,103,playback(no-service) thank you -- Atif

hello there, I have searched lists about an application chan_spy, people talked about it on lists that we can use it to monitor sip to sip calls. but I am unable to find any clue of it. can some one please tell me from where I can get this chan-spy application thank you regards, -- Atif

...f there is any other application to hang up the destination channel, what is that? Also what is the status of originating channel? Where should the call to second AGI be put in the dial plan? I hope you guys understand my problem/issue. Please guide me, thanks alot in advance. -- Best Regards Atif Razzaq http://atif-razzaq.blogspot.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110329/d04089f8/attachment.htm>

Hello everyone, I''m wondering what the current status of hardware performance counter usability in Xen is. I see some old posts describing the diffculties of virtualizing hardware counters within dom0 and the domUs, but not much else. Have they been implemented or are they in the process of being implemented? Or are there no future plans for implementation? Any help would be

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

...bscript: Implemented Oct. vbscript: Fixed Hex. Stefan D?singer (15): ddraw: Update the palette before presents to the NULL window. ddraw/tests: The testbot applies color keying without a key. ddraw/tests: Correct executebuffer offsets. wined3d: Improve color fixups in atifs shaders. wined3d: Check for conversion changes in the atifs fragment pipeline. wined3d: Add per-context private data for fragment pipelines. wined3d: Avoid constant collision in atifs. ddraw/tests: Make sure color keying is on in test_texturemapblend. ddraw/tests: Por...

...and 4 bit alpha formats to WINED3DFMT_A8_UNORM. wined3d: Add an explicit break in case of unhandled BUMPENVMAP. ddraw: Create a dynamic buffer if DDLOCK_DISCARDCONTENTS is used. wined3d: Bind the src in a manual presentation blit. wined3d: Correctly count used stages in the atifs pipeline. wined3d: Test the correct program for native limits. wined3d: Use sign fixup for the atifs bumpenv matrix. wined3d: Add GL_ALPHA to the atifs argument replicator debug function. d3d9/tests: Skip some texture transform tests if shaders are unsupported. d3d9/t...

...there is a way to change the xen bridge (vif) of a running domain. For example, if I have booted a domU with xenbr1 as vif. Now is it possible to change the vif to xenbr2 while the domU is running? If yes, then can someone guide me a little so as to where to start looking? Best Regards, Muhammad Atif __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hello All, We are looking for a SIP provider teminating calls in India, Pakistan and Bengladesh. Any one knows a good one? Regards, Cesar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040824/4d5dd4b4/attachment.htm

hello there, any one who used ASTCC in a real enviroment, or has successfully handled above 1k simultanous calls. need some evalution of ASTCC. if any one has such an experience please share it with the rest thank you Atif

...nel to both parties where as eth1-IP is a private IP and call can't send rtp on eth1-IP. h323 has to send eth0-IP in mediaControlChannel to the caller. if we bind h323 to 0.0.0.0 as we do in SIP, it sends 0.0.0.0 as it is to the caller and callee. your help will be appriciated Regards, -- Atif

Hi, I am trying to install asterisk on Suse 9. I downloaded asterisk-1.2.1.tarand untar this package. I am following the README and the installation instruction to run "make" ans "make install". But I can not find any "make" or "make install" in the directory asterisk-1.2.1. Can any one please help me how can I install asterisk-1.2.1 on Suse? What am I