search for: sgupta

...Buildbot has detected a new failure of llvm-i686-linux on llvm. Full details are available at: http://google1.osuosl.org:8011/builders/llvm-i686-linux/builds/3952 Buildbot URL: http://google1.osuosl.org:8011/ Buildslave for this Build: dunbar1 Build Reason: Build Source Stamp: 78383 Blamelist: sgupta BUILD FAILED: failed svn sincerely, -The Buildbot

...86-linux on llvm. > Full details are available at: >  http://google1.osuosl.org:8011/builders/llvm-i686-linux/builds/3952 > > Buildbot URL: http://google1.osuosl.org:8011/ > > Buildslave for this Build: dunbar1 > > Build Reason: > Build Source Stamp: 78383 > Blamelist: sgupta > > BUILD FAILED: failed svn > > sincerely, >  -The Buildbot > > > _______________________________________________ > LLVM Developers mailing list > LLVMdev at cs.uiuc.edu         http://llvm.cs.uiuc.edu > http://lists.cs.uiuc.edu/mailman/listinfo/llvmdev >

...are available at: >> http://google1.osuosl.org:8011/builders/llvm-i686-linux/builds/3952 >> >> Buildbot URL: http://google1.osuosl.org:8011/ >> >> Buildslave for this Build: dunbar1 >> >> Build Reason: >> Build Source Stamp: 78383 >> Blamelist: sgupta >> >> BUILD FAILED: failed svn >> >> sincerely, >> -The Buildbot >> >> >> _______________________________________________ >> LLVM Developers mailing list >> LLVMdev at cs.uiuc.edu http://llvm.cs.uiuc.edu >> http://lists.cs.u...

...The Buildbot has detected a new failure of llvm-ppc-linux on llvm. Full details are available at: http://google1.osuosl.org:8011/builders/llvm-ppc-linux/builds/2002 Buildbot URL: http://google1.osuosl.org:8011/ Buildslave for this Build: nick1 Build Reason: Build Source Stamp: 73491 Blamelist: sgupta BUILD FAILED: failed svn sincerely, -The Buildbot

..., Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 10000 T305 Timer: 30000 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 However, when we try and make a call: -- Executing Dial("SIP/sgupta-901d", "ZAP/1/32953295") in new stack Mar 24 11:11:31 NOTICE[987]: app_dial.c:746 dial_exec: Unable to create channel of type 'ZAP' == Everyone is busy/congested at this time Any ideas? Cheers, Sahil

Hi, I'd like to know how I can playback a pre-recorded message to a user using our system without answering the call. I want to do the above in the scenario where the user dials a number and the number has been dialled incorrectly. Regards, Sahil Gupta VoiceValley

Hi, I have a Gateway running in "TE" (terminal equipment mode as "slave" that I need to connect to my asterisk server using a TE100P card. Can anybody give a quick run up of how to run the TE100P's in Network Termination mode to have this working sucessfully? Cheers! Regards, Sahil Gupta VoiceValley

Hi, We seem to be having some teething issues with a new Hylafax - happy to pay someone to complete the installation. Please contact offlist. Regards, Sahil Gupta Chief Executive Officer VoiceValley Group of Companies Phone: +61-7-30188403 Fax: +61-7-30188499

...uiuc.edu] On Behalf Of Chris Lattner > Sent: Saturday, November 01, 2008 1:49 AM > To: Commit Messages and Patches for LLVM > Subject: Re: [llvm-commits] [llvm] r58505 - > /llvm/trunk/lib/VMCore/Type.cpp > > On Oct 31, 2008, at 3:26 AM, Sanjiv Gupta wrote: > > > Author: sgupta > > Date: Fri Oct 31 05:26:20 2008 > > New Revision: 58505 > > > > URL: http://llvm.org/viewvc/llvm-project?rev=58505&view=rev > > Log: > > For some targets pointer and int are 16-bits. Allow 16-bits as a > > valid index > > in such cases. > &...

Hi, does anyone know the prices for g729 codecs from Digium? I sent an email a while ago to them but haven't got any response so far. Prices are per unit or volume? Thanks, -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

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Hi, We seem to be having an interesting issue with Asterisk whereby, it keeps routing calls coming in to the 'default' context.... regardless of what changes occur to h323.conf. <SNIP> [POP-A] type=user host=1.2.3.4 context=international </SNIP> == Starting H323/ip$1.2.3.4:12914/16313 at default,12126599878,1 failed so falling back to exten 's' == Starting

Hi, Is there anybody on the list that recommends anyone for colocation/telehousing in the US? I'm after 2 Servers to be hosted in the US, preferably on the west coast. Regards, Sahil Gupta VoiceValley

Hi, We are terminating around 60 channels on one of our Asterisk boxes, which the client sends in H323 mode. Client (MERA) --> H323 --> Asterisk --> IAX --> Asterisk The problem we face is that at random intervals the H323 process (as part of Asterisk) dies and can no longer accept new calls whilst Asterisk is still running happily. We have to then kill asterisk and start it

Hi, I am currently interconnecting to a PRI using a Cisco AS5350. I'd like to be able to dial specific numbers out by a specific isdn channel, so for e.g. if I dial 999 01 12341234 it should send 12341234 out via isdn channel one from the Cisco AS5350. If somebody would be able to guide on this, it would be appreciated. Regards, Sahil Gupta VoiceValley

Hi, I'm looking for someone to do time-to-time mantainence on some of our machines going up in New York. The person *MUST* be stationed in New York. Areas of expertise required: - Proficiency in Linux: Slackware, Fedora - Proficiency on Cisco Routers If anybody is interested, please contact me off-list. Regards, Sahil Gupta VoiceValley

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi,I am looking for a reliable provider that can provide 3 dedicated linux servers asap. Unfortunately, the provider I have used for YEARS has become way too slack in recent times and we have to move on. Cheers, Sahil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081114/e06232f4/attachment.htm

Hi, I am trying to change the dialplan to enable call recording (incoming and outgoing calls) on the "click of a button". Is it possible? All the documentation I found so far, enable recording for 'all calls' to an extension. Does this code look ok? Currently Recording "on" only for 1030 when user presses *44, start recording. *55 to stop recording