similar to: i am missing something!

hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck my configs are. sip.conf [general] port=5060 bindaddr=192.168.10.195 disallow=all allow=alaw allow=ulaw [101] username=101 type=friend secret=1234 host=192.168.10.195 context=sip callerid="101"<101> defaultip=192.168.10.176 extensions.conf [globals] [incoming] exten =>

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

Hello. I am trying to find a way to retrieve data from Harvard Dataverse website. I usually don't have problem in web-scraping data but the problem here is that there are a bunch of data formats such as .tab, .7z and so and I just can't find a way to retrieve the data I am interested in woth an unique solution. Any hint? [[alternative HTML version deleted]]

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, I just started with Asterisk as I am very unhappy with the functionality of my current PBX at home. I try to understand everything and play around, but it is not as easy as I thought. So please be patient if this is a too easy question for You. I installed Asterisk 1.4.26.3 on a Debian Lenny with IP 192.168.2.147 My extension.conf looks like this: [default] exten => 1001,1,Answer()

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

Hi, I have a weird problem woth the time zone on my virtual (goDaddy) dedicated server I had to setup the timezone using system-config-date to GMT+7 i am in NY (GMT-5) when i do GMT-5 i get all the times 2 hours ahead seems like there's something im missing here If i add an event to my web app let's say 7:30 - 8:15 , the only way for me to show it right is GMT+7 can anyone help me

Hi I've a problem with a gnugkv2.0.7 I've compiled gnugk successfully I've installed PWlib-1.6.6 and openh323-1.13.5 libraries successfully When i run gnugk i have this error: error while loading shared libraries liboh323_linux_x86_r.so.1.13.5 cannot open shared object file No such file or directory I try to use command export: export

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

I'm trying to upgrade a very old sparcstation running Solaris 8 which is running dovecot 1.x for few users. All I have for the task is good old gcc 2.95.2. The poor sod complains because it can't compute the sizeof(unsigned char prefix_text[]) at line 13 of log-error-buffer.c. Can I help it by - say - putting a constant between the '[]'? Or is it unfair? :-) I don't think

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: > >