similar to: SIP INFO request in asterisk

Hello, I have one question regarding *. Default configuration for asterisk is to keep configuration(s) in ordinary text based config files. My question is simple: is it possible to keep those config info (at least, to start from - sip.conf, extensions.conf and voicemail.conf) on a database, which asterisk access via ODBC. If it is possible, I'd appreciate if someone points me where I can

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

Is there a way to display registered SIP useragents and sort them from CLI? I.N.

Hi. We have an odd problem with incoming SIP calls. I have attached a SIP debug log, with some asterisk verbosity as well, demonstrating the problem, below. Is this a known bug? Vital stats: - Asterisk 1.2.3 - Sipura SPA-841, SPA-941 phones - Fedora core 3 The problem manifests itself with these symptoms: - an internal SIP extension receives a call from our PRI - the SIP phone answers the

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two

Hi David, > The code above is needed as the GHC calling convention redefines what > registers are considered callee save. No one else rummages in to the > original function as all the other calling conventions use the same > set of callee and caller save registers, so GHC is the only one that > needs to differentiate. shouldn't the caller also know what registers are callee

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

What would be the best way to solve this problem? Anyone else that have got the same problem with Android’s native SIP client, especially on Samsung phones? I do not know if the bug is in Android native SIP, or Samsung’s build of the SIP client, or if the bug is even with the OpenVPN client, or where the bug actually is. The ACK might even be sent for real, but have the incorrect source IP so

Hi, Using SIP methods and options, is there any way for a callee to send the caller a short text message when the call is establishing ? Scenario is : Alice and Bob's SIP phones are registered to an Asterisk server. Alice calls Bob : an INVITE message is sent to Bob's phone Bob is replying : a 200 OK message is sent back to Alice with a short text included ("Welcome to BoB

> Every inlinable call in a function that has debug info (F->getSubprogram() returns non-null) must have a DebugLoc associated with it that has a scope chain that ends in that same DISubprogram. Thank you for the comment! I don't know if this is a proper way to fix, but after I add DebugLoc same as inserting position instruction, no error occurs.

Hi Duncan, Any word on this making 3.0? Cheers, David On 16 October 2011 23:03, David Terei <davidterei at gmail.com> wrote: > On 15 October 2011 00:31, Duncan Sands <baldrick at free.fr> wrote: >> Hi David, >> >>> The code above is needed as the GHC calling convention redefines what >>> registers are considered callee save. No one else rummages in

On 15 October 2011 00:31, Duncan Sands <baldrick at free.fr> wrote: > Hi David, > >> The code above is needed as the GHC calling convention redefines what >> registers are considered callee save. No one else rummages in to the >> original function as all the other calling conventions use the same >> set of callee and caller save registers, so GHC is the only one

Hi Duncan, Karel, On 14 October 2011 08:22, Duncan Sands <baldrick at free.fr> wrote: > Hi Karel, > >>>> > const unsigned* >>>> > ARMBaseRegisterInfo::getCalleeSavedRegs(const MachineFunction *MF) >>>> const { >>>> > + bool ghcCall = false; >>>> > + >>>> > + if (MF) { >>>> > + const

Every inlinable call in a function that has debug info (F->getSubprogram() returns non-null) must have a DebugLoc associated with it that has a scope chain that ends in that same DISubprogram. https://llvm.org/docs/SourceLevelDebugging.html discusses some of the debug info IR metadata in LLVM. On Fri, Feb 2, 2018 at 1:03 AM Ku Nanashi via llvm-dev < llvm-dev at lists.llvm.org> wrote:

Hi. I have a beginner conceptual question about Asterisk: Let's suppose that there are 4 softphones registered in my Asterisk and all of them are currently online. In addiction , there is no call. Suddenly, one of these softphones sends a SIP message to the Asterisk. In this case the dialplan will execute the instruction ' exten => 2005,1,Dial(SIP/2000&SIP/2001&SIP/2002,

I've got a very strange problem and I can't figure it out. I have a Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I see callerID name, but it is not getting to * via SIP. I am running * 1.4.2 and the latest Cisco IOS for my router. Here is what is happening: A call comes into the gateway. It sends a SIP INVITE to * with "pending" as the callerID

Hi Karel, >>> > const unsigned* >>> > ARMBaseRegisterInfo::getCalleeSavedRegs(const MachineFunction *MF) >>> const { >>> > + bool ghcCall = false; >>> > + >>> > + if (MF) { >>> > + const Function *F = MF->getFunction(); >>> > + ghcCall = (F ? F->getCallingConv() == CallingConv::GHC : false);

LLVM developers, I was wondering if the program would still be safe if I strip the byval attributes from the parameters in the entire bitcode. LLVM language reference manual states that "The attribute implies that a hidden copy of the pointee is made between the caller and the callee, so the callee is unable to modify the value in the callee. This attribute is only valid on LLVM pointer