similar to: One-way talk paths (without INVITE?) and other issues

I just downloaded the latest CVS. RTP streams from a X-Lite on a PC, to the Linux box running Asterisk and Linphone, seem to connect: > -- Called m4 > -- SIP/m4-dd8f is ringing > -- SIP/m4-dd8f answered SIP/m12-3649 > -- Attempting native bridge of SIP/m12-3649 and SIP/m4-dd8f But then Linphone sends an endless stream of complaints: > (linphone:20223):

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

--T4sUOijqQbZv57TR Content-Type: text/plain; charset=us-ascii Content-Disposition: inline Dear R, I would like to report what I think is a bug in R. I am running R within emacs on a Digital AlphaStation. See the version information at the end of my R session for details. I also attach a copy of the file that is read in the `read.table' command. Here's my R session, with a few

dear expeRts, i would like to merge the data frame entries in a list. for example: > #input > myl <- list(q1=data.frame(id=c("Alice", "Bob"), grade=c(90, 49)), q2=data.frame(id=c("Alice", "Chuck"), grade=c(70, 93)), q3=data.frame(id=c("Bob", "Chuck"), grade=c(84, 40))) > #output > (mydf <-

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

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>

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

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, 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

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 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

Hi Conrad, Thanks for your help. I am unlucky. No i have this following error : source='speexdec.c' object='speexdec.o' libtool=no \ depfile='.deps/speexdec.Po' tmpdepfile='.deps/speexdec.TPo' \ depmode=gcc3 /bin/sh ../depcomp \ arm-linux-gcc -DHAVE_CONFIG_H -I. -I. -I.. -I../include -I../libspeex -I/usr/include -O2 -fno-exceptions -O2

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

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 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

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

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 make[2]: quittant le r?pertoire ? /usr/src/linphone/arm/speex-1.1.6/src ? make[1]: *** [all-recursive] Erreur 1

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