Re: [Sofia-sip-devel] Crash when system time jumps

My patch to fix this issue: commit f50f04be51f0e43a20ac2dceef1a8115d76dbb25 Author: Michael Jerris m...@jerris.com Date: Fri Mar 28 14:33:47 2014 -0400 FS-6356: --resolve fix assert when you set the time 15 days in the future during operation while things are in queue to be processeed

Re: [Sofia-sip-devel] Crash when system time jumps

...@yahoo.at wrote: On 06/12/2014 02:30 PM, Michael Jerris wrote: + /* check it there are still things queued, if there are, that means everything scheduled is 15 days in the future */ +/* in this case, we had a large time shift, we should schedule for 15 days in the future (which

Re: [Sofia-sip-devel] obtaining sip message as string

The only way I know of is out of the log callback from tport_log On Jul 24, 2014, at 11:41 PM, Dave Horton d...@dchorton.com wrote: Hopefully this is a quick question for someone I've been using nta and dealing with the parsed message structures that it provides. But now I have a need

Re: [Sofia-sip-devel] any way to bind client to specific interface(eth/wlan).

I do. I think I am the only one still around that does. All that being said, i think its probably time we make a real fork of this if no one from the original group is going to be working on it any more. Is there anyone here that is willing to dig though our patches on freeswitch and figure

Re: [Sofia-sip-devel] Administrator needed

And I am back! Sorry for the delay. I do have access to gitorious... but its probably time to really fork this. We can probably move off of what we have in the freeswitch repo,but I'll need a volunteer to pick through the patches we have added in our tree, and see if we have hard-coded

Re: [Sofia-sip-devel] Sofia-sip

Kai! long time no talk. I'm fine with whatever folks want, just need a little help as far as manpower to clean up what we have back to something usable outside freeswitch. Mike On Aug 10, 2014, at 11:13 AM, Kai Vehmanen kvli...@nosignal.fi wrote: Hi, [resend to list] On Sat, 9 Aug

Re: [Sofia-sip-devel] Sofia-sip

I wouldn't plan on any, as i said, the next step is a volunteer to review the changes made in the freeswitch tree so we can re-release that as something others can use without breakage. On Aug 15, 2014, at 8:39 AM, Thomas Volkert si...@gmx.net wrote: Hello, Great! For a long time, I am now

Re: [Sofia-sip-devel] How to change media mode in active call?

I know it works with USER_SDP_STR, what exactly is your call to update/invite? are you doing ACTIVE_AUDIO(1) ? On Sep 3, 2014, at 9:13 AM, Andreas Wehrmann a.wehrm...@yandex.com wrote: Hi! I'm somewhat at a loss here. I'm in a SIP call that I made or accepted with one of

Re: [Sofia-sip-devel] Help me, example code

sofia-sip is a sip stack, not an rtp stack. you can see examples in sofia-sip-cli and freeswitch. Mike On Dec 24, 2014, at 6:15 AM, Андрей Аппалонов appalonovand...@gmail.com wrote: Hello! Please tell me where I can take the example of the establishment of the channel (the call is

Re: [Sofia-sip-devel] Help!

This is not a minor feat. We have already created this functionality inside of FreeSWITCH. Maybe its a better fit for your needs. On Dec 29, 2014, at 7:13 AM, Андрей Аппалонов appalonovand...@gmail.com wrote: Hello! I am writing an application using the library Sofia-SIP. The

Re: [Sofia-sip-devel] nta outgoing create: invalid URI

this issue sounds really familiar but I looked yesterday and could not find where we hit it or fixed it in the past. Just to eliminate could you try the Sofia-sip code from inside the FreeSWITCH codebase and see if it resolves this issue or not, we have a variety of patches in that codebase On

Re: [Sofia-sip-devel] nta outgoing create: invalid URI

PES > > Product Engineering Services > Software | Embedded | Semiconductor Frost & Sullivan Company of the Year > 2013-2014 > > > -- > *From:* Michael Jerris <m...@jerris.com > <javascript:_e(%7B%7D,'cvml','m...@jerris.com');>>

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

I'm interested On Monday, February 15, 2016, Antonis Tsakiridis < antonis.tsakiri...@telestax.com> wrote: > Hello, > > I just wanted to let you know that we 've been working with Sofia SIP and > successfully combined it with Google's WebRTC native library for iOS and > lately we managed to

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

gt; > Best regards, > Antonis > > On Thu, May 19, 2016 at 6:31 PM, Michael Jerris <m...@jerris.com > <javascript:_e(%7B%7D,'cvml','m...@jerris.com');>> wrote: > >> The closest to a central maintained repo at this point is the copy in the >> freeswitch tree.

Re: [Sofia-sip-devel] Sofia SIP integration with WebRTC & BoringSSL for iOS

o > do so. Do you have auto tools know-how? If so would you be interested in > contributing that part your self so that the community can benefit from a > more generic solution? > Best regards, > Antonis > > On Mon, Feb 15, 2016 at 5:31 PM, Michael Jerris <m...@jerris.com >

Re: [Sofia-sip-devel] call-ID troubles

Callid in sdp? On Saturday, May 14, 2016, Francesco Lamonica wrote: > Hi all, i know this might be a shot in the dark however... > i am experiencing a strange issue, randomically on a UAC that uses > sofia-sip (latest stable from repository) when putting the remote

Re: [Sofia-sip-devel] call-ID troubles

o Michael, > sorry, obviously i meant SIP :) > > > On Sun, May 15, 2016 at 12:02 AM, Michael Jerris <m...@jerris.com > <mailto:m...@jerris.com>> wrote: > Callid in sdp? > > > On Saturday, May 14, 2016, Francesco Lamonica <alienpeng...@gmail.com > <

Re: [Sofia-sip-devel] Sofia SIP 'master' repo

Its still on my list…. but still not done. Antonis, yeah, lets coordinate and see if we can figure out what needs to be done to make it happen. Mike > On Jul 17, 2017, at 9:14 AM, Antonis Tsakiridis > wrote: > > Ok thanks Moises for the useful feedback. I

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