similar to: Still no audio on SIP phone

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

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 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

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 ------------------------------------------------------------------- |

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....

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

Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is there any way to configure X-lite to use the headset instead of the speakers? Or will I have to plug the headset in the speaker jack ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

-- 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:

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

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

Hi I installed Asterisk on WhiteBox Enterprise Linux 3.0 respin 1 The process of installation was the following: First I compiled and installed Zaptel, in order to have ztdummy (uncommented in Makefile). I loaded the ztdummy (modprobe ztdummy) and then i installed Asterisk: make make install make configuration make samples I started Asterisk, and created one SIP account, with the following

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

I've been looking at Asterisk for a replacement for our phone system and I'm hoping someone can help validate my assumptions. We have 4 analog lines coming into the building. These lines are simple POT lines and we have them in a "hunt group" with Verizon so that when a single phone number is dialed, the first line is rang, if that line is busy it will ring the second line, and

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

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

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go

I just installed Xorcom Rapid and I'm trying to connect with Xlite. In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address of the new install. I can ping that box. When I try to connect I get hung on the "Awaiting Proxy login information" and the log reads: ======================================================================== ? 2004 Xten Networks, Inc. All

Hi list, I'm trying, as usual, to set up voicemail. It works, but signaling to phones, doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:voicemail@mydomain.com Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:.