similar to: Best software SIP client

Has anyone got the BT Speedway (AVM Fritz) card working on a RedHat 8.0 system with *. If so could someone give me some pointers on getting the right sequence of installing the drivers and which versions to use. Thanks, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been using X-Lite on FWD without any troubles and recently became interested in trying asterisk. I am able to register from X-Lite and dial a number - I've tried dialing some of the sample numbers in the sample extentions.conf file, like 500 and 1234, they appear to dial correctly from X-lite but nothing else happens - no audio is heard. My understanding is that I should hear some

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Which rpm has libstdc++.so.5? xten-xlite for linux says it needs this. Of course, there will be something else it will need after I get this....

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guies, I know I know this subject have been The most written subject about VoIP Right... but I just want to make clear, just one time ! If Asterisk is on a Public IP Address and a softphone behind the nat, sip.conf must contains for this phone: nat=yes .... Now if I want to configure my sipphone (X-Lite) placing behing the NAT, it must have in "Domain/Realm" the external IP

Hello everyone. I am having problems getting speex support. It seems * is not loading speex. When i did a make in the codecs sub dir, the following error pops up when making speex: codec_speex.c:34:19: speex.h: No such file or directory is this file missing in the cvs as i just removed the whole * dir and did a new checkout and still seem to get this error, or do i need to get/install

Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. Thanks in advance, Hamish ------------------------------------------------------------------- |

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards with POTS have no such issues. Any clue where the

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

hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some

Has anyone had any success using the Florz patch for zaphfc ? I have a * system with 2 HFC cards which is working fine with 2 PTP ISDN lines however the users are complaining of crackles on the line which I am assuming is related to the IRQ issues raised by Florz. I have tried to use the patch but it errors trying to patch zaphfc.h Any help would be appreciated. Regards, Stuart -- No virus

Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

I am having a problem getting music on hold working one of my servers. I have had this working on a PII 400 just fine but decided to upgrade my Asterisk server to a PIV 1.5ghz. I have installed mpg123 which seems to be working fine but when I start *, I get the following error message at the CLI prompt when I start *: WARNING[81931]: File chan_oss.c, Line 232 (sound_thread): Read error on

Hi, I'm currently looking for a softphone for windows, we have been using X-Pro but it appears that X-pro doesn't support Message Waiting notification. Does anyone know of a well featured softphone that does support MWI ? I can't seem to find one. Any suggestions would be most appreciated, Thanks, Simon Ward

Hi, I have a polycom reboot script which sends a NOTIFY with check-sync. It worked fine with 1.3.4. After I upgrade to 1.4.1, it stopped working. Anyone has the same problem? Thanks, Richard

Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users

Hi all, Seriously, I've tried to read everything I could find (& search for) on voip-info.org and other sites about this problem, but have been unsuccesful. Equipment: xten lite X100P Whitebox linux running Asterisk / AMP D-Link DI-804HV (VPN router) I have installed another DI-804HV at a second location and created a tunnel. For the computers behind that unit, everything works fine