similar to: asterisk-users Digest, Vol 37, Issue 88

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

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

I used this site (and perhaps a couple other Google returned) as well as the Polycom Admin guide as reference. http://www.voip-info.org/wiki/view/Getting+MWI+on+Polycom+Phones+to+work+wit h+Asterisk 299 is the extension one dials to access their voicemail (with caller ID sending to the correct voicemail). I see the bold on the wiki page telling me to put in my voicemail context, but I'm

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"

Hi All, I have a Polycom 501 that is behind a NAT. When it registers to the Asterisk server it is using the IP address on the private network and not the public IP of the NAT address. Can someone tell me what I need to do so the phone registerers using an internet address rather than the remote network NAT address. -------------- next part -------------- An HTML attachment was

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 <

> James Harper wrote: > > >Additionally, just to satisfy myself that I wasn't going mad I changed > >the port from 5060 to 5070 and now things are working, so something is > >definitely playing up on port 5060. > > > >James > > > > > > > You probably have are behind NAT and your NAT device has a SIP ALG. > Changing the port disables

Hi, Some IP Phones (Aastra) are able to send a custom HTTP request just after registration completion. Using this, it is possible to update phone's screen with messages like "Do Not Disturb" or "Forwarded To VM". RFC 3680 (http://www.faqs.org/rfcs/rfc3680.html) provides a mecanism to support these interactions. To my knowledge, this RFC is not implemented yet in

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 All; I succeeded to have a success call from Polycom behind NAT while Asterisk has public IP address, but I was not able to have a succeed call (it was established, but no voice running, and then the call disconnected) if Asterisk behind NAT and Polycom behind NAT. When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

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,

The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't know where to look. p From: "Von L." <methodvon@gmail.com> To:

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

Hi, All, I'm trying to build a customized backend and I need to lower the formal arguments like this: There are several specific registers just for storing formal arguments. And also there are several general purpose registers for computation. If there is an instruction which uses parameters, I should first use a move instruction, which moves the value to general purpose register. For

nat is equal to yes, and server definition in ftp provisioning server is correct. i followed packets between phone and asterisk, it seems for some reason asterisk is not happy about challenge response its getting from polycom. why its not happening in the same LAN, beats me! and also NAT device is cisco SIP aware and works flawlessly with Linksys. the only close definition of issue i have found is

Hi all- Equipment: Xlite softphone Polycom 301 with SIP 2.1.1 and BootROM 3.2.3 Polycom 501 with SIP 2.1.1 and BootROM 3.2.3 Asterisk 1.4.2 SIP Trunk to FWD I wanted to post this problem as I haven't found it described in any of the past presence threads on here. I use an identical configs for a Polycom 501 and 301. (I actually unplug one when the other is in use). The