similar to: Add opus codec for Linphone on Ubuntu

Hello I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the console version of linphone. both boxs are using the latest alsa drivers on a LFS kernal 2.4. I am running into errors with codec compatability between linphone and *. A point to note is that I am able to connect to asterisk using other sip phones noteably sjphone however linephone is giving me

It seems a new Linphone 4.2 is to be published next week ! Hopefully, ... Le ven. 5 juin 2020 à 13:34, John Hughes <john at calva.com> a écrit : > On 26/05/2020 15:33, Olivier wrote: > > Hi John, > > 1. Could you get any further, in your quest for working BLF with linphone ? > > The patches to get linphone-3.12 BLF working with Asterisk are here: > >

On 26/05/2020 15:33, Olivier wrote: > Hi John, > > 1. Could you get any further, in your quest for working BLF with > linphone ? The patches to get linphone-3.12 BLF working with Asterisk are here: http://perso.calvaedi.com/~john/linphone-3/ They're pretty damnned trivial: 1. add the "Accept" header to the SUBSCRIBE message so asterisk doesn't reject it. 2.

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Today's "frustrated programmer" award goes to Linphone, which has the following debug output: > (linphone:28655): LinphoneCore-WARNING **: this fucking remote sip phone did not answered properly to my sdp offer! I get this message when I connect to linphone using a softphone, or when I try to use linphone to connect to asterisk and listen to an announcement. I suspect that

Hello. Calls cancelled by caller during the dialing phase, are shown in Linphone as simply past calls, not missed ones. I thought this is an Linphone issue, but Sylvain says it's on my PBX side: https://github.com/BelledonneCommunications/linphone-android/issues/832#issuecomment-557474864 > The CANCEL message has a Reason header with Q.850 protocol and cause 0, which doesn't mean

Hi, I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get linphonec to work with Asterisk. I have the echo test working, but when I dial in to this, to voicemail or anything else using Playback() to play a sample, I hear nothing for ages (10-15 secs) and then little sections. With the echo test, I get the tail of the message (...pressing the pound

On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com> wrote: > So I've got a bit further with my project to get BLF working between > asterisk and linphone. > > Initially asterisk was rejecting linphone's SUBSCRIBE messages because > they didn't have an Accept: header. I've fixed that and now the initial > SUBSCRIBE messages work and I see all my

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

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

> On 23/03/2020 18:51, Joshua C. Colp wrote: >> On Mon, Mar 23, 2020 at 2:45 PM John Hughes <john at calva.com >> <mailto:john at calva.com>> wrote: >> >> >> Why is asterisk giving an error 500? I can find no reason, there >> is nothing in any log. >> >> >> The sequence number is from the past. The first SUBSCRIBE is

Hi John, 1. Could you get any further, in your quest for working BLF with linphone ? 2. Have you tried with a different Linphone version (4.12 is pending on Linux, packaged as an AppImage, or 4.11 exists on iOS/Android/Win10) ? Best regards Le mer. 25 mars 2020 à 15:06, John Hughes <john at calva.com> a écrit : > > On 23/03/2020 18:51, Joshua C. Colp wrote: > > On Mon, Mar

So I've got a bit further with my  project to get BLF working between asterisk and linphone. Initially asterisk was rejecting linphone's SUBSCRIBE messages because they didn't have an Accept: header. I've fixed that and now the initial SUBSCRIBE messages work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and

Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the

Hey Guys, I've setup Asterisk on a PC here in the office. I installed the sample configs. With a the sound card connected to some speakers and launching asterisk with the "-vvvc" command line, I can issue the "dial" command and hear the playback of the recorded message. When connecting to the asterisk server from my PC using Linphone I connect to 500@hostname. The

Hi Sergio On 16.10.20 at 07:54 sergio wrote: > Sometimes, linphone shows missed calls as missed. Look like asterisk > replies with cause=487 that time, but I can't understand why. > > Grandstream always shows calls as missed ones. > > How should I investigate this? You could try to reproduce it while activating pcap traces and analyze it afterwards - or you could

Hi All, I am getting this error during make. please help me./ speexec.c: In function `speex_ec_process': speexec.c:112: syntax error before "noise" cc1: warnings being treated as errors speexec.c:133: warning: implicit declaration of function `speex_echo_state_reset' speexec.c:148: warning: passing arg 5 of `speex_echo_cancel' makes pointer from integer without a cast

Hello, you can compile opus with "--enable-fixed-point". That saves a cpu ressources. https://github.com/Studio -Link/PKGBUILDs_clean/blob/master/opus/PKGBUILD#L23 -- nice regards, Sebastian Reimers ------------------------------------------ IT-Service Sebastian Reimers Am blanken Boom 14 32369 Rahden Festnetz: 05776-3930000 Fax-Nummer: 05221-17242088 Skype: miete-admin E-Mail: