search for: linphonec

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

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 cour...

...sifting through the archives. Q1 what is the dtmfmode based on ( i read that it is determined according to the phone being use and i have also read that it is base on he codec being used, rfc2833 for alaw/ulaw info/outband for others. Does this need to be set for me to carry out a test from linphonec to * default message? Q2 the rtp ports used by asterisk rang from 10000/20000 while linphone uses 7078, which I changed to 17078 to allow it to fall within the range. however I notice during the invite/ok trans. that rtp uses 18XXX and then uses 0 when no codec can be agreed upon. Q3 Does...

I am dying here with linphone (not sure if it is crap software or just me being an idiot) but out of the box debian installations of two linphones fail with a "Got SIP response 415 "Unsupported Media Type" back from 192.168.1.3" Can anybody recommend a particular SIP soft phone that broadly satisfies the following criteria? 1. Run on linux. 2. Simple to use and setup. 3. Is

...ux.org/frs/?group_id=7&release_id=209 Documentation: http://docs.blackfin.uclinux.org/doku.php?id=bf537-stamp_board Open Software: - Linux for Blackfin: http://blackfin.uclinux.org/ General Documentation http://docs.blackfin.uclinux.org/ Pre-compiled image, supporting Linphonec http://blackfin.uclinux.org/frs/?group_id=17&release_id=233 Documentation for Linphone (using Speex) http://docs.blackfin.uclinux.org/doku.php?id=linphone Patches for Linphonec, to use LCD & Keypad: http://blackfin.uclinux.org/pm/task.php?func=detailtask&project_task...

Hi Sorry if this comes twice; i sendt first version from non-member address. I'm learning use Asterisk but cannot solve following problem: i have Asterisk v1.0.7 (DEbian) and Linphonec v1.2 (Debian). Every time i try to register within LAN i got 'Forbidden' message from Linphonec. Where to start searching for reason for this failure, is there more debuggin options available? Here is relevant part of sip.conf: [fujitsu] type=friend username=arimo context=siptest se...

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal. Could anyone of you p...

I'd like to cobble together a videophone from an analog phone, connected to an Asterisk FXS channel, and a co-located video camera, connected to a video grabber card on the Asterisk server (so I have a Linux video device providing the video stream). When a call is made from the phone, I'd like to somehow add the video and produce a SIP video call. I don't want to use any sort of

I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm

Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/8bc76bbb/attachment.htm

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 key........p...

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem