similar to: IAX media

Displaying 20 results from an estimated 4000 matches similar to: "IAX media"

2004 Jul 19
2
callparking vs calltransfer
HI ALL; Anybody can explain the difference between "call parking " vs "call transfer" Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040720/2975991b/attachment.htm
2005 Jul 06
4
converting windows .wav to .gsm
HI ALL; I have problem converting a windows .wav file to .gsm format by Sox. Could anyone help. Cheers, Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3408bfd5/attachment.htm
2004 Jul 07
1
OH323-COMPILE
HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,
2007 Apr 29
1
Voicemail Creation
HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee -------------- next part -------------- An HTML attachment was
2008 May 18
1
Bridging a call on hold with an active call
Dear All I want to use asterisk for the following Senario and Need help to find a SAMPLE extension.conf Incoming call >>>>>>>>>>>Asterisk >>>>>>>>>>>>>>GSM Termination Gw first leg second leg What I want to do is putting first call leg on
2004 Aug 02
9
asterisk+radius
HI ALL; Is there anybody who use app_radius(astersik radius module)??????????? is it stable? Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040803/8a096bfe/attachment.htm
2004 Dec 25
1
Alert-Info
Hi; Any idea of how to have different ringing tone on called party for different caller-id by means of "Alert-Info" header. Regards Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041226/3611eeae/attachment.htm
2004 Jul 19
2
codec translate
HI ALL; Is astersik enable to translate between different codecs. I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa. Regards mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 Mar 09
2
Asterisk-oh323-0.7.1 compile error
Hi; I use the following asterisk, openh323, pwlib: asterisk = cvs-head-03/09/05 openh323 = 1.13.5 pwlib = 1.6.6 asterisk-oh323= 0.7.1 Asterisk, openh323, pwlib were compiled successfully but when I try to compile Asterisk-Oh323-0.7.1 , I got the following error: chan_oh323.o chan_oh323.c chan_oh323.c:37:34: asterisk/channel_pvt.h: No such file or directory ......... ...........
2004 Jul 22
0
Re: h323ep----gnugk-----astersik------h323ext
HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native
2004 Nov 29
1
IAX port
HI ALL: I am newbie to IAX, my iax.conf is as follows: [general] port=5036 ..... but I donot why it doesnot listen on UDP PROT 5035, instead it listens on 4569 Asterisk CLI debug says: [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Nov 30 11:52:12 WARNING[1076220544]:
2004 Jul 26
1
voicemail+g729
HI ALL; I found in the following page: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing 1-If I could record all IVR promts in G729 format 2-If I could record voicemail in g279 format with """format_g729.c""""" then I donot need any g729 license (I suppose all my clients have g729 ip phones) My question is, how
2014 Jan 17
2
[LLVMdev] [icFuzz] Help needed with analyzing randomly generated tests that fail on clang 3.4 trunk
Hi Hal, Just submitted 27 failing tests on clang version 3.5, trunk 199158. http://llvm.org/bugs/show_bug.cgi?id=16431 I expect that these failures correspond to 2+ unique bugs. Cheers, -moh -----Original Message----- From: Hal Finkel [mailto:hfinkel at anl.gov] Sent: Thursday, January 09, 2014 6:01 AM To: Haghighat, Mohammad R Cc: llvmdev at cs.uiuc.edu Subject: Re: [LLVMdev] [icFuzz] Help
2013 Jul 29
2
[LLVMdev] [icFuzz] Help needed with analyzing randomly generated tests that fail on clang 3.4 trunk
----- Original Message ----- > Hal, > > Just posted a package containing 214 small tests showing bugs in the > latest Clang (3.4 trunk 187225) on MacOS X when compiled at -O2. > http://llvm.org/bugs/show_bug.cgi?id=16431 > > These are new tests different from the previously posted ones, but > their root causes could be the same as before or could actually be > new
2013 Jul 29
0
[LLVMdev] [icFuzz] Help needed with analyzing randomly generated tests that fail on clang 3.4 trunk
Hal, Just posted a package containing 214 small tests showing bugs in the latest Clang (3.4 trunk 187225) on MacOS X when compiled at -O2. http://llvm.org/bugs/show_bug.cgi?id=16431 These are new tests different from the previously posted ones, but their root causes could be the same as before or could actually be new bugs. Cheers, -moh -----Original Message----- From: llvmdev-bounces at
2017 Jun 18
1
Extremely slow du
Hi Mohammad, A lot of time is being spent in addressing metadata calls as expected. Can you consider testing out with 3.11 with md-cache [1] and readdirp [2] improvements? Adding Poornima and Raghavendra who worked on these enhancements to help out further. Thanks, Vijay [1] https://gluster.readthedocs.io/en/latest/release-notes/3.9.0/ [2] https://github.com/gluster/glusterfs/issues/166 On
2013 Jul 27
2
[LLVMdev] [icFuzz] Help needed with analyzing randomly generated tests that fail on clang 3.4 trunk
Hal, I ran the failing tests in the attachment to the bug 16431 on the latest clang trunk (version 3.4 trunk 187225). http://llvm.org/bugs/show_bug.cgi?id=16431 The following tests still fail: Tests in diff: t10236 t12206 t2581 t6734 t7788 t7820 t8069 t9982 All tests in InfLoopInClang: t19193 t22300 t25903 t27872 t33143 t8543 Meanwhile, I'll launch a new run
2017 Jul 11
2
Extremely slow du
Hi Kashif, Thank you for your feedback! Do you have some data on the nature of performance improvement observed with 3.11 in the new setup? Adding Raghavendra and Poornima for validation of configuration and help with identifying why certain files disappeared from the mount point after enabling readdir-optimize. Regards, Vijay On 07/11/2017 11:06 AM, mohammad kashif wrote: > Hi Vijay and
2005 May 20
1
RDNIS (DNID) Call Routing
I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)?
2007 Apr 30
0
voicemail + Dynamic mailbox
HI All; I want to use Asterisk for just Voicemail Server and I need a Dynamic creation of Mailboxes. My users 's Mailboxes are same as "Extensions" but I donot want to add mailboxes in "Voicemail.conf" Is there any way to create mailbox from Asterisk dial-plan ? Appreciate any suggestions Mohammad Mirzaee Mohammad Mirzaee -------------- next part -------------- An HTML