Re: [Asterisk-Dev] Sending a REINVITE to a SIP channel

This is for a connection that is already established. I receive a connection, once I detect it's a fax I want to redirect it to another SIP server via a REINVITE. Will "Dial" work for this? Does it really issue a REINVITE? I thought that Asterisk remained between both endpoints. On Fri May 27

Re: [Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

On Sat, 2005-05-28 at 10:56 +1200, Derek Smithies wrote: > Does anyone have some realworld experience on a recent machine & software > timers to prove they are still a bad idea? Anyone using ztdummy on a 2.6 kernel is using precisely the same software timers which drive the kernel's POSIX timer s

Re: [Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

Hi, > >software timers are lacking. > > > > > > I think that is just narrow thinking. Steve's original link for why software timers are a bad idea was from 2002. Since then we have had fantastic improvements in a) cpu performance b) latency of the kernel - anyone tried the low la

Re: [Asterisk-Dev] Sending a REINVITE to a SIP channel

Juan Jose Comellas wrote: For some applications I'm building on top of Asterisk sometimes I need to redirect a SIP channel to another server. I intended to do this via a REINVITE, but I haven't found an easy way to do this from an Asterisk application. Is there any way to do this? What about

[Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

Using zaptel just for a time source is silly. We should use POSIX timers instead. Here's a start, but I don't know how long it'll be before I can get to finish it, so I thought I'd post it here in the hope that someone else will pick it up. This demonstrates how to set up POSIX timers to interrupt

Re: [Asterisk-Dev] why chan_sip:regcontext registers Noop ?

On Fri, May 27, 2005 at 08:49:37AM -0700, Kevin P. Fleming wrote: > Luigi Rizzo wrote: > > I notice that chan_sip.c has the option, when given a regcontext > > argument in sip.conf, to register an entry in the dialplan of the form > > > > exten => 3456,1,Noop(3456) > > Right, so in that conte

Re: [Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

On May 27, 2005 12:32 pm, Tilghman Lesher wrote: > On Friday 27 May 2005 05:33, David Woodhouse wrote: > > Using zaptel just for a time source is silly. We should use POSIX > > timers instead. Here's a start, but I don't know how long it'll be > > before I can get to finish it, so I thought I'd pos

Re: [Asterisk-Dev] why chan_sip:regcontext registers Noop ?

Luigi, Take a look at the SIP Registration section of the following patch: It allows you to do execute any dialplan function when a SIP or IAX device registers or the register expires. This adds the functionality that the regcontext doesn't have. -C

Re: [Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

Tilghman Lesher wrote: On Friday 27 May 2005 05:33, David Woodhouse wrote: Using zaptel just for a time source is silly. We should use POSIX timers instead. Here's a start, but I don't know how long it'll be before I can get to finish it, so I thought I'd post it here in the hope that someon

Re: [Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

Tilghman Lesher wrote: On Friday 27 May 2005 05:33, David Woodhouse wrote: Using zaptel just for a time source is silly. We should use POSIX timers instead. Here's a start, but I don't know how long it'll be before I can get to finish it, so I thought I'd post it here in the hope that someone el

Re: [Asterisk-Dev] High resolution timers using POSIX clocks instead of zaptel

On Friday 27 May 2005 05:33, David Woodhouse wrote: > Using zaptel just for a time source is silly. We should use POSIX > timers instead. Here's a start, but I don't know how long it'll be > before I can get to finish it, so I thought I'd post it here in the > hope that someone else will pick it up

Re: [Asterisk-Dev] why chan_sip:regcontext registers Noop ?

Luigi Rizzo wrote: I notice that chan_sip.c has the option, when given a regcontext argument in sip.conf, to register an entry in the dialplan of the form exten => 3456,1,Noop(3456) Right, so in that context, you have also: exten => _3XXX,2,Dial(SIP/${EXTEN}) Using 'regcontext' allow

Re: [Asterisk-Dev] Incorrect behavior : address for SIP responses

I am not very familiar with the SIP standards, however the behavior you describe seems to be legit. http://www.faqs.org/rfcs/rfc3581.html When the client sends the request, if the request is sent using UDP, the client MUST be prepared to receive the response on the same IP address and po

[Asterisk-Dev] Re: [Asterisk-Users] does Jitter calculation in chan_iax2.c work???

SteveK and Andrew,    Thanks a lot for the suggestion. It helped. We didnt know that jitterbuffer wont be enabled with sip endpoints. "forcejitterbuffer=true" solved the problem. Thanks again, Vijay & AshishOn 5/27/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: On May 27, 2005 01:47 am, V

Re: [Asterisk-Dev] does Jitter calculation in chan_iax2.c work???

On May 27, 2005, at 1:47 AM, Vij wrote: Hi, We are trying to get the jitter of a channel for iax channels. iax2 show netstats The above command always shows zero value for jitter. (Actually, only rtt and kpkts are non-zero). The behaviour is the same even for cross-continental calls.

[Asterisk-Dev] Incorrect behavior : address for SIP responses

Hello all, When receiving a SIP request, I've observed that asterisk SIP channel returns responses to the address it was received from. For exemple, if a SIP UA, name it A, sends an Invite from a port UDP: to asterisk port UDP:5060, then Asterisk replies a 100 Trying to A:! On so on for ea

[Asterisk-Dev] why chan_sip:regcontext registers Noop ?

I notice that chan_sip.c has the option, when given a regcontext argument in sip.conf, to register an entry in the dialplan of the form exten => 3456,1,Noop(3456) where 3456 is the extension number. I don't understand how is this supposed to be used. In all examples I saw, and in most app