similar to: IAX transfer bug in last CVS ?

Hi Folks, A couple of questions about making SMB (3 or 4) authenticate to an external (anonymous) LDAP server: 1) A typical LDAP user record is below. Is there anything lacking in this record that would prevent Samba from authenticating against our LDAP server? Note the sambaSID is as is, gobblygook info: dsAttrTypeNative:eduPersonAffiliation: Employee Member dsAttrTypeNative:givenName: David

Hello All When I open up iaxcomm, it registers fine with the asterisk server. If I call into it, iaxcomm will ring; however if I leave iaxcomm sitting idle for awhile (I haven't figured out exactly how long) it seems to miss calls. I can see the calls coming in on the asterisk server but they never ring through on iaxcomm. If I close it and reopen it, it takes calls again fine. I thought I

Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

Talking to myself ... ;-) Solved this by ... disallow=all allow=gsm ;allow=ulaw ;allow=alaw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paulo Mannheimer Sent: quinta-feira, 11 de dezembro de 2003 09:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hi, I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as

Hello, When i have a "Hangup" in my dialplan (extensions.conf) the RFC says to terminate the session is to send a BYE request to UA. After tracing the traffic on port 5060 UDP i recognized that my asterisk is NOT sending a BYE request to it's peer, so the peer doen't know to end the session and continues to send RTP packages to me. Does anyone know how to fix this? Here's

Hello, How can I get peerip, recvip, from, uri, useragent, peername, t38passthrough variables in (within) my custom Asterisk application? I can't use chan_sip.c internal structures (such as sip_pvt) in my custom application, because there's no chan_sip.h and I can't include it into my application (maybe there's other way?). I can do like this: exten =>

iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome.

Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=...

Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, Tim --

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

Hi, I seem to be having problems with IAX clients based on the iaxClient library. I have been working on my own client (an augmentation to the Call Manager I released last week) and it seems to regularly miss incoming calls entirely. It also occasionally misses the drop signal when the remote end drops a call. Has anybody else seen this kind of behavior? I have tested with my client, with

I'm trying to set up * server behind NAT. The box is set up as DMZ in my DSL router, i.e. all incoming connections without explicit port mapping are forwarded to *. So far I'm unable to get this setup to work for either IAX or SIP (tried IAXComm & XLite softphones on public IP address). Data seems to come in fine (IAX/SIP debug shows message interaction taking place), but there is no

Hello, I am having horrible echo problems when using the FireFly client on both the caller and callee sides of the call. When I use another IAX soft client like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone else experienced this and do you know what might be the problem? Thanks, dj -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 05/17/2018 11:38 AM, Frank Vanoni wrote: > On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote: > >> 3. How do I set up the server to block these ? >> >> 4. Can I stop the retransmitting of the 401 Unauthorized packets ? > > I'm happy with Fail2Ban protecting my Asterisk 13. Here is my > configuration: > > in /etc/asterisk/logger.conf: > >

Hmm. By the way, please don't post bugs to asterisk-dev as I've been told :> That list if for on-going development. That sounds like a bug I encountered in 1.0.5. There is a division by zero bug in chan_iax2.c introduced somewhere after 1.0.4 I believe and currently fixed in HEAD. (They've given me enough shit for posting the bug while it was fixed in HEAD already. No need to

iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol. It is distributed as part of Steve Kann's iaxclient library. I've just posted new Windows, Linux and Mac OSX binaries to sourceforge. The Windows binary was compiled on WinXP. The Linux binary was compiled on RedHat 9. The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4 (Tiger) beta.

On advice from others I dropped gnophone in favor of iaxComm. I am operating on an IBM T30 laptop Redhat Linux 2.4.20-8 with an Intel i810 audio chipset (comes in the laptop). I am using the Gnome desktop. There is no reference to alsa or oss to be found. All audio components function fine. Nothing else is running and I have an active broadband internet connection. I can ping www.digium.com