similar to: Polycom 320 messages

All, I'm using Asterisk 1.6 and using Polycom 500's with SIP firmware 2.1.3. I can not seem to get the Message Waiting Indicator to work reliably (and in my opinion correctly) with voicemail. I've got the following in my phone.cfg: <reginfo> <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.callBack="*97"

While I know this is not a true asterisk problem, I figure someone where may know. When you click on Messages and it gives you the count of Urgent, New, etc. How can you make the phone gather that information? For example, my phone shows me there is an e-mail. It also sends an e-mail. Yet, when I click on message before I connect to the contact center, it doesn't have any counts. Here is

Don, All you need to do is configure your VM button on the phone via the cfg files. You setup the phone to look to an FTP server for config scripts and then setup the scripts to work as needed. Here is an example of where you set this VM button from my file called <mac address>-phone.cfg... <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="1006"

I am running CentOS 5 with Asterisk 1.4.14. I am trying to setup storage of voicemail messages into MySQL. It is my understanding that I can only do this via ODBC. I installed per http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured and built Asterisk, using menuconfig to

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

I think Polycom has added another feature that nobody wants. With MWI configured, and a phonexxx.cfg that has this: <msg msg.bypassInstantMessage="1"> <mwi msg.mwi.1.subscribe="" msg.mwi.1.callBackMode="contact" msg.mwi.1.callBack="XXX" msg.mwi.2.subscribe=...> </msg> Under 1.3.4 and earlier, the phone would immediately

Hi I am just getting started with asterisk and trying out using the sample files with a Polycom IP500 (latest sip.ld etc) My question: phone status says not registered: /system status/server status "Line 1 poly is not registered" console message says: handle_request: Registration from '<sip:poly@10.1.20.3>' failed for '10.1.20.50' I have tried dialing and

First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

On Jun 30, 2008, at 9:25 AMPDT, Duncan Sands wrote: > Hi Dale, yes it's just stack alignment. Unfortunately > your patch doesn't make any difference. > > Thanks for thinking about this, > > Duncan. OK, the test is not quite right then. This doesn't happen on Darwin so I'm afraid you'll have to debug it. Perhaps testing for

Hi Dale, > OK, the test is not quite right then. This doesn't happen on Darwin > so I'm > afraid you'll have to debug it. Perhaps testing for > (RegInfo->needsStackRealignment(Fn) && > FFI->getObjectIndexEnd()!=0) > ? yes, that works - thanks! Since I have no idea what this is doing, is it ok if I leave it to you to commit it?

On Jun 27, 2008, at 3:11 AMPDT, Duncan Sands wrote: > FAIL: test/CodeGen/X86/vec_shuffle-10.ll > Failed with exit(1) at line 3 > while running: llvm-as < test/CodeGen/X86/vec_shuffle-10.ll | llc - > march=x86 -mattr=+sse2 | not grep {sub.*esp} > subl $16, %esp > subl $16, %esp > child process exited abnormally Is the code actually loading and storing

Hi Dale, yes it's just stack alignment. Unfortunately your patch doesn't make any difference. Thanks for thinking about this, Duncan. On Friday 27 June 2008 19:51:00 Dale Johannesen wrote: > On Jun 27, 2008, at 3:11 AMPDT, Duncan Sands wrote: > > FAIL: test/CodeGen/X86/vec_shuffle-10.ll > > Failed with exit(1) at line 3 > > while running: llvm-as <

Looks good. + unsigned temp; + if (is64bit) + temp = RegInfo.createVirtualRegister(&PPC::GPRCRegClass); + else + temp = RegInfo.createVirtualRegister(&PPC::G8RCRegClass); How about? const TargetRegisterClass *RC = is64Bit ? &PPC:GPRCRegClass : &PPC:G8RCRegClass; unsigned TmpReg = RegInfo.createVirtualRegister(RC); Evan On Jul 9, 2008, at 9:16 AM, Gary

Hi, RFC3680 defines a SIP event package for registration. This event package which can be used through NOTIFY-SUBSCRIBE methods, seems very useful for free sitting or presence applications. This package is supported in various SIP phones (at least Thomson ST2030) : when turned on, this feature adds a new login/logout menu among other things. It can also be used to send Welcome notices to mobile

Hello, after some time I'm trying to build my code with the current CVS of LLVM, and have a problem. The mentioned file, around line 184, contains: if (FixedSlot == FixedSpillSlots+NumFixedSpillSlots) { // Nope, just spill it anywhere convenient. FrameIdx = FFI->CreateStackObject(RegInfo->getSpillSize(Reg)/8,

On 12/07/2018 03:36 PM, Administrator TOOTAI wrote: > Le 07/12/2018 à 14:32, hw a écrit : > > [...] >> >> Queues seem to be the only way to have several phones ring at once, or >> are there other ways? > > Dial(SIP/Phone1&SIP/Phone2&...&SIP/Phonex,,) > Good to know, thanks! What are the entries needed in the queue_members table when using

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any*

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Hello, it seems the Prolog/Epilog insertion does not correctly align stack for me. Consider the PEI::calculateFrameObjectOffsets method. It only aligns the stack if FFI->hasCalls() is true. The only place where MachineFrameInfo::setHasCalls is invoked is PEI::saveCallerSavedRegisters and the value 'true' is only passed when there are instructions with opcodes equal