similar to: canreinvite problems

Hi, I'm experiencing an issue with my x86_64 machine containing a Hauppauge PVR-500 (ivtv) and a Digium TDM400p (wctdm, part of zaptel) PCI cards. Independently of each other both cards work fine, but once the wctdm driver is loaded and mythtv tries to record something on the PVR-500 the wctdm driver freaks out. The error message is see is "TDM PCI Master abort" printed over and

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being cancelled. No problems with other phones (polycom, grandstream). Attached the complete sip debug log

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

Maybe there's an ntp expert out there who can help me with this. I have an NTP server serving our local network. It is set up to use pool.ntp.org servers for it's upstream sync. ntpq -p reveals that the server is stuck on stratum 16, which I understand means "not synced". The clients are unable to sync with my local server because of this. Here's the output of ntpq -p

I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:

Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines.... You could read the complete Press Release here:

Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in

We just released a new version of the idefisk iax2 softphone, version 1.21 beta, available for download at http://www.asteriskguru.com/tools/idefisk_beta.php Some bugs were fixed, some new bugs might have been introduced :) - The problem with delays is finally gone!!! (one of the bugs was a memory leak, everybody using an older version is encouraged to upgrade.) Privacy Warning: Version 1.21 of

Hello everyone, I have a problem when loading the library: IlluminaHumanMethylation27k.db I installed the package from zip (from the Bioconductor website), and also installed all the packages it asked for. It went all succesfull: "package 'IlluminaHumanMethylation27k.db' successfully unpacked and MD5 sums checked" But when I am trying to load this library I get an error:

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@<my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

We just uploaded the latest and greatest version of the idefisk iax2 softphone, version 1.24 Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us

Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten => *601,1,Answer exten => *601,n,Dial(IAX2/pbxnetwork/xxxxxx,30,m) exten => *601,n,Hangup exten => *602,1,Answer exten => *602,n,Dial(IAX2/pbxnetwork/xxxxxx,30) exten => *602,n,Hangup No I have a problem when

Tofik Suleymanov wrote: > Steve Totaro wrote: > >> Give idefisk a try. It works very well for me, its free, and does >> not crash all the time like Cubix (formerly Firefly). >> >> > > Hello Steve, > > As far as i know 'idefisk' is a softphone, but i need a hardware phone. > Thank you for reply. > > Tofik Suleymanov > Oooops, sorry its

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 > Message: 13 > Date: Tue, 24 Jun 2003 13:12:45 +0200 > From: Patrik Gustavsson PS Sweden Senior Technical Consultant > <Patrik.Gustavsson@Sun.COM> > Subject: [Samba] Has anybody used the mysql passdb backend in samba > 3-beta1 ? > To: samba@lists.samba.org > Message-ID: <3EF8322D.4040204@sweden.sun.com> >

2009/2/5 Olivier <oza-4h07 at myamail.com> > Hi, > > Here http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38 is a > table listing ATA/Gateways combinations. > Could anyone successfully set a Patton M-ATA to work with another one, > using Asterisk 1.4 ? > > Is reinvite (canreinvite=yes) necessary or not ? > > Regards > > Replying to myself, I

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assump??o Loureiro msn: loureiro_bruno@hotmail.com

Hello, I have a Sangoma A101 connected to an ISDN30 (E1 in the UK) with some Snom 300 and Idefisk softphones. I can do SIP and IAX2 calls just fine, however I cant get any audio in either direction on the Zap channels. When I call in or dial out over the ISDN30 (UK E1) I can see the call answered/placed on the CLI and then silence follows. I've been provisioned 25 out of the 31 channels only

Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server? Amaury Rodr?guez http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 Go OpenSource And Be FREE!! --------------------------------- Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze.

Hey guys, I'm having some trouble with gain settings using a Wildcard X100P and Zaptel 1.4 Beta 2 on Asterisk 1.4 beta 3. My transmit volume between an IAX client (idefisk) and a POTS device through my 1.4 box is great, but my receive volume is terrible. I can hardly hear a word they say. I've cranked my gains up to 100, inserted the wcfxo module with boost=1, but haven't had