similar to: Differents config files

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hi! When I try to make a call with ohphone, that is the message I get: Call to "06302" aborted, insufficient bandwidth Can anybody tell me a solution or a reason why this messages appears? Thanks a lot! Regards, Mireia

Hi! I have been looking for a while for informatoin about how QoS is assured in Asterisk, but I haven't found a thing. Can someone give me some tips about that? Thanks, Best regards, Mireia

Hi all! I am looking for some free software to monitoring all the calls that are being done in my network. Which telephone are connected, how long are the calls, quality of service, bandwidht,etc. If someone knows about a good one, plesea tell me. Regards, Mireia

Hi all! I am using ohphone (well, I am trying to) to make calls. I will make an H.323 - SIP Gateway but I don't understand the architecture of all this. What is the exact role of asterisk? It can be used as gateway, that I know, but what else can he do? Is it necessary to have ohphone to make calls or asterisk can also do that? So when the gateway it is going to be implemented how is it

Hi all! One easy question... I hope someone will answer me. I've installed asterisk with the samples. Somewhere in my network I have an H.323 Gatekeeper. What must I do to make that the gatekeeper talk with Asterisk? And I another little question... with the samples installed asterisk works ok? What must I install to see how it works? I am lost!!!!!!!!!! Please help me! See you. Mireia

Hi! I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a gateway between this network and the SIP network. Now I can do calls from de foreign network (SIP) to the locla (H.323) but I don't know how to do the inverse. The H.323 terminals use NetMeeting, and when I try to make a call, it says that the number dialed must be registered in the gatekeeper. How can I register

Hi all! Please I need someone that have already done an H.323 - SIP gateway to help me with some problems. I can stablish calls from a SIP telephone to a H.323, but I can't do vice versa... (problems with port 1719- when the gatekeeper tries to contact with asterisk at this port, it is unrecheable...). Please someone can help me? Regards, Mireia

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-?-vis. I know there is an option available for the Avaya systems but it?s a little out of the price range I?m looking for (~$200/channel). Has anyone out there found a stable way to do this?

Was in the process of upgrading a 7960 to SIP and in advertently applied a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then goes on to repetedly ask for SEP<MACaddr>.cnf. Help!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I have pass operating on bitcode file which produces more than one equivalent representation. opt --my-pass <%s | Filecheck %s --my-pass generates files a.rpt b.rpt c.rpt . How do i write test without writing all 3 files to stdin. I have considered CHECK-LABEL for each. it creates bulky checks. Thanks Mahesh -------------- next part -------------- An HTML attachment was

That works. Thanks. One more followup question though. Once i run opt on bitcode, there is not useful output/transform on bitcode. this rpt files are extra. I am hoping to do something like this, ; RUN: FileCheck --input-file=a.rpt.gold --check-prefix=CHECK-A < a.rpt ; RUN: FileCheck --input-file=b.rpt.gold --check-prefix=CHECK-B < b.rpt i did not find much examples in tests hence

Hi all, I am trying to convert a data set of physician death codings (each individual's cause of death is coded by multiple physicians) from long to wide format, but the "reshape" function doesn't seem to work because it requires a "time" variable to identify the sequence among the repeated observations within individuals. My data set has no order, and different

Hi Mahesh, On 31 March 2018 at 11:45, Mahesh Attarde via llvm-dev <llvm-dev at lists.llvm.org> wrote: > opt --my-pass <%s | Filecheck %s > > --my-pass generates files a.rpt b.rpt c.rpt . How do i write test without > writing all 3 files to stdin. You can run FileCheck over them on separate RUN lines assuming you know the filename (which I assume you do since you'd be

I've installed fail2ban, it seems to be working as it identified my failed test logins, BUT, my question is: what can I do when I see same invalid name trying to login to dovecot, different IP each time, how can I say block each IP as used by this name ? or it that a bad idea ? I can see two persistent attempts as so: I don't have such user 'ignacio' or 'julian' # grep

See: http://llvm.org/docs/CommandGuide/FileCheck.html It is not required to pipe output to FileCheck; there is the --input-file option, which allows you to run FileCheck on an existing disk file. Something like this: FileCheck %s --input-file a.rpt --check-prefix=A FileCheck %s --input-file b.rpt --check-prefix=B FileCheck %s --input-file c.rpt --check-prefix=C If there are common parts to each

Oops. My bad. I mean to write match-file being my match pattern. and a.rpt being file generated by opt. ;RUN: FileCheck <match-file>a.rpt.gold --input-file=a.rpt. mahesh On Sat, Mar 31, 2018 at 6:57 PM, Mahesh Attarde <coder.mahesh at gmail.com> wrote: > That works. Thanks. > > One more followup question though. > Once i run opt on bitcode, there is not useful

Folks, I've just successfully set up Asterisk (as part of the Asterisk Management Portal installation). When I say "successfully", I mean that I have gone through all the steps detailed for the installation of AMP and not hit any snags there. I can connect to my asterisk server via ssh and can also connect via Http to the portal to change settings in AMP. Now I'm trying to