Re: [asterisk-dev] ast_sip_session

2016-11-15 Thread Kelvin Chua
Forgot to clarify, I am also trying to pass some headers on response, i.e 180, 183 Kelvin Chua On Wed, Nov 16, 2016 at 12:22 PM, Kelvin Chua <kel...@gmail.com> wrote: > thanks for the insights, i'm doing this as a proof of concept. > i relaized there are a lot of things which

Re: [asterisk-dev] ast_sip_session

2016-11-15 Thread Kelvin Chua
Wed, Nov 16, 2016 at 1:40 AM, Matthew Jordan <mjor...@digium.com> wrote: > > > On Tue, Nov 15, 2016 at 6:17 AM, Joshua Colp <jc...@digium.com> wrote: > >> On Mon, Nov 14, 2016, at 03:46 AM, Kelvin Chua wrote: >> > playing around chan_pjsip/res_pjsip >> >

[asterisk-dev] ast_sip_session

2016-11-13 Thread Kelvin Chua
but stuck on how to write this to the reply downstream. any pointers? Kelvin Chua -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-dev] new module

2015-07-27 Thread Kelvin Chua
Thanks mark, will try this out Kelvin Chua On Wed, Jul 22, 2015 at 10:45 PM, Mark Michelson mmichel...@digium.com wrote: On 07/21/2015 11:23 PM, Kelvin Chua wrote: i am building a proof of concept for a new module, my first step was to create a socket to listen to, accept connections

[asterisk-dev] new module

2015-07-21 Thread Kelvin Chua
(desc.local_address, bindaddr); ast_tcptls_server_start(desc); return AST_MODULE_LOAD_SUCCESS; } question: what would be the general flow of session_do() in this scenario? Kelvin Chua -- _ -- Bandwidth and Colocation

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-30 Thread Kelvin Chua
Works alex, i added a couple more parameters to copy from the original Invite fmtp. hope this find its way into trunk soon. Kelvin Chua On Mon, Jun 29, 2015 at 9:58 PM, Kelvin Chua kel...@gmail.com wrote: alex, i'll try it out, will give you feedback tomorrow. Kelvin Chua On Mon, Jun 29

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-29 Thread Kelvin Chua
alex, i'll try it out, will give you feedback tomorrow. Kelvin Chua On Mon, Jun 29, 2015 at 9:47 PM, Kelvin Chua kel...@gmail.com wrote: yes matt, i have it loaded Kelvin Chua On Mon, Jun 29, 2015 at 8:32 PM, Matthew Jordan mjor...@digium.com wrote: On Mon, Jun 29, 2015 at 4:36 AM

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-29 Thread Kelvin Chua
yes matt, i have it loaded Kelvin Chua On Mon, Jun 29, 2015 at 8:32 PM, Matthew Jordan mjor...@digium.com wrote: On Mon, Jun 29, 2015 at 4:36 AM, Kelvin Chua kel...@gmail.com wrote: Guys, just tried asterisk13 and added seanbrights' patch for opus. incoming INVITE has fmtp

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-29 Thread Kelvin Chua
because it is just a passthrough. have you changed anything to chan_sip.c to make this work? Kelvin Chua On Sat, Jun 27, 2015 at 7:26 AM, Kelvin Chua kel...@gmail.com wrote: i verified parse_sdp is doing its job correctly and stores it in struct. but after going back to chan_sip somehow somewhere

[asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-26 Thread Kelvin Chua
to get away from this behavior and act like a proxy and just copy the fmtp from the ingress? Kelvin Chua -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-26 Thread Kelvin Chua
i verified parse_sdp is doing its job correctly and stores it in struct. but after going back to chan_sip somehow somewhere everything resets before generate_sdp. maybe because i am working on ast12, i'm going to try 13 On Jun 26, 2015 5:49 PM, Joshua Colp jc...@digium.com wrote: Kelvin Chua

Re: [asterisk-dev] storing INVITE fmtp and use it to send relay

2015-06-26 Thread Kelvin Chua
oh that figures. i am working on ast12 and never bothered trying 13. i'm playing around with opus and so frustrated that it does not adapt to the incoming sdp. even if after parse_sdp, it just resets everything on generate_sdp. i'll try 13 later, thanks for the tip On Jun 26, 2015 5:30 PM,

Re: [asterisk-dev] rtptimeout

2015-01-28 Thread Kelvin Chua
Hi Matthew, you are right, digging around testing and found out this broke rtptimeout Set(JITTERBUFFER(adaptive)=150,,30) for reasons I haven't found out yet Kelvin Chua On Tue, Jan 27, 2015 at 11:34 PM, Matthew Jordan mjor...@digium.com wrote: On Mon, Jan 26, 2015 at 8:22 PM, Kelvin Chua

Re: [asterisk-dev] rtptimeout

2015-01-28 Thread Kelvin Chua
to work this way correct? Kelvin Chua On Wed, Jan 28, 2015 at 4:41 PM, Kelvin Chua kel...@gmail.com wrote: Hi Matthew, you are right, digging around testing and found out this broke rtptimeout Set(JITTERBUFFER(adaptive)=150,,30) for reasons I haven't found out yet Kelvin Chua On Tue, Jan

[asterisk-dev] rtptimeout

2015-01-26 Thread Kelvin Chua
Hi guys, I noticed rtptimeout on asterisk 12 is not working, so i looked at the source. looks like, it has no effect on res_rtp_asterisk? Kelvin Chua -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [Asterisk-Dev] OMG THE SKY IS FALLING!! NOT!!!

2004-05-15 Thread Kelvin Chua
yah, it's just a matter of just being there with a pair of wires and tap it in! what voip needs right now is sip over tls and srtp :) On Fri, 2004-05-14 at 23:17, Steve Totaro wrote: I disagree about the snooping thing. I have been in hundreds of phone closets and a great many have been

Re: [Asterisk-Dev] OMG THE SKY IS FALLING!! NOT!!!

2004-05-15 Thread Kelvin Chua
how about the cross-talks? without even exerting any effort to listen-in you get to hear the lastest gossip the neighbors are discussing over their twisted pairs... On Sat, 2004-05-15 at 02:57, Steven Critchfield wrote: On Fri, 2004-05-14 at 13:29, Jennifer Archer wrote: I think land lines are