similar to: Newbie

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 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

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

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 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, in these dark days of COVID-19 lockdown I'm using linphone to connect to my office asterisk system for working from home. It's going pretty well but the presence/BLF functions don't appear to work. In the linphone logs and asterisk debug I find that asterisk is rejecting linphone's PUBLISH message: <--- SIP read from UDP:10.27.128.3:5060 ---> PUBLISH sip:john at

Hi As per the Linphone, Readme.arm I tried to compile the speex. -------------------------------------readme.arm-------------------------------------------------- ........... Cross compiling speex for ARM: ******************************** First you need to remove ogg headers from your build system to avoid a dirty conflict between your build machine binaries and the arm binaries. They

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

All the clients that I'm aware of for IP telephony have drawbacks. Some won't work at all. KPHONE -- Kphone works best for me, but Kphone doesn't have a dialpad to dial tones during the middle of the call, so the demo that * comes with can't be run. Kphone (3.1, the latest) also has a habit of crashing if you do something even mildly stressful, such as hang up while Kphone is

Hello ppl, At initial level i configure asterisk woth only soft phones ,in which one at windows machine and other is linux i am using windows messenger and linphone respectively both phones registered with asterisk respectively problem is that they bypass asterisk on call when i send request from linphone to messenger request shown on messenger but on asterisk console nothing to and also if i send

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

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after

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, 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, this problem seems to be coming up a lot lately -- the README.arm in linphone tells people to remove libogg headers, but speexenc and speexdec require them to build. Perhaps we need a README.arm in libspeex telling people not to remove the libogg headers. K. On 01/04/2008, Erwan A <mout551 at hotmail.fr> wrote: > > Hi all, > > I am following the README.arm from Simon

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: > >

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

On 01/04/2008, Erwan A <mout551 at hotmail.fr> wrote: > > Hi, > > Yes i agree with you. You don't have to delete these files. > > But if i cross compile with ogg header files, i have the following error : > > > /usr/lib/libogg.so: could not read symbols: Invalid operation > collect2: ld returned 1 exit status > make[2]: *** [speexenc] Erreur 1

Hello, I'm trying to compile linphone 3.0.0 in Ubuntu And I'm getting the following compilation error. I somewhere read that the fault happens because of the gcc 4.3.2 that debian based linux uses. Is there any solution (except downgrading gcc) Thank you in advance if /bin/bash ../libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I.. -I../include/ -I.. -I../../oRTP/include

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic