search for: sflphoned

I want to set sflphone as extension on asterisk. I have a sip account/DID with vitelity.net. Not sure what to put in the wizard: alias ??? hostname ??? is this the asterisk server hostname, or the hostname where my sflphone is sitting on the lan (it's a home network) username ??? is this the assigned extension number? password ??? is this the assigned extension number password? Any

...ice quality (mic). I tried Sflphone (has alsa support) and that worked on FC4 (not on FC3) and it does not work on Centos44 either: [root at raaf ~]# rpm -i sflphone-qt-0.6.2-1.i386.rpm error: Failed dependencies: libstdc++.so.6(GLIBCXX_3.4.4) is needed by sflphone-qt-0.6.2-1.i386 sflphoned is needed by sflphone-qt-0.6.2-1.i386 yum provides libstdc++.so.6 libstdc++.i386 3.4.6-3.1 installed Matched from: /usr/lib/libstdc++.so.6 [root at raaf ~]# ll /usr/lib/libstdc++.so.6 lrwxrwxrwx 1 root root 18 Mar 9 05:56 /usr/lib/libstdc++.so.6 -> lib...

Hi all, Can some one suggest me an IAX client for Linux and Windows? I used KIAX once, but know it seems complicated to have it working on Ubuntu. Thanks. Ronaldo.

hi all, can anybody help me to give Opensource SIP client information which can be modified as per our requirment regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090518/802cc3ac/attachment.htm

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

I think I'm able to connect with Ekiga, at least it reports "registered". Curiously, when I exit Ekiga and switch to SFLphone, it isn't able to connect with the exact same parameters; it just says "trying" and never resolves. I'm not able to test outside connectivity because of too many hops: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:thufir

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

are there any open source sip softphone (Window OS version )?

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Processing commands for control at bugs.debian.org: > severity 777787 serious Bug #777787 [src:autofs] autofs: ftbfs with GCC-5 Severity set to 'serious' from 'important' > severity 777871 serious Bug #777871 [src:geis] geis: ftbfs with GCC-5 Severity set to 'serious' from 'important' > severity 777912 serious Bug #777912 [src:insighttoolkit]

Processing commands for control at bugs.debian.org: > severity 777785 important Bug #777785 [src:attica] attica: ftbfs with GCC-5 Severity set to 'important' from 'normal' > severity 777787 important Bug #777787 [src:autofs] autofs: ftbfs with GCC-5 Severity set to 'important' from 'normal' > severity 777871 important Bug #777871 [src:geis] geis: ftbfs