There are some other clients, even if they are mainly testing/demo applications for some SIP stacks.
sofsip-cli for SofiaSIP (which is backed by Nokia) simpleopal for OpalVOIP They do work, even if they're not as full featured as linphone in some ways, e.g. on soundcard management. They offer some more options in other fields. Regarding the Playback issue, it seems that Playback into a [ConfBridge|MeetMe] conference stutters and drops randomly. I think I'll file a bug for that. Thank you Il 12/11/2010 10:23, Sebastian ha scritto: > Hi > > On 11/11/2010 03:35 PM, Matteo Fortini wrote: > >> 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 2.x and 3.x. >> >> I just need a console scriptable softphone, so maybe there's an >> alternative to linphone (which seemed good enough anyway!)... >> > I use linphonec as well - and haven't found another console sip phone > either. I'd be interested if there is another one. > > Sebastian > > >> Thank you, >> Matteo >> >> > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
