search for: multiprotocol

Displaying 20 results from an estimated 29 matches for "multiprotocol".

2020 Jul 02
0
Multiprotocol File Sharing via NFSv4 and Samba
Hi Kraus, I know that Gluster can be exported over NFS-Ganesha (supports v4.X), Samba (protocol 1.0 in order to get 'real' permissions), Apple's stuff and if you rebuild the source - you can use the built-in gNFS (supports NFS v3 over tcp) all at once. Yet, I'm not sure about the ACLs, so you should either test it yourself or ask on the gluster mailing list. Deployment
2020 Jul 02
5
Multiprotocol File Sharing via NFSv4 and Samba
Hi all, are there any non-commercial solutions (apart from solutions like Dell EMC, IBM and NetApp) around that allow to simultaneously access the same file system via NFSv4 and Samba exports in a (nearly) non-conflicting manner, especially w.r.t. to NFSv4/Windows ACL incompatibilities? Best Sebatian ____________________ Sebastian Kraus Team IT am Institut f?r Chemie Geb?ude C, Stra?e des 17.
2020 Jul 02
1
Multiprotocol File Sharing via NFSv4 and Samba
FreeNAS / FreeBSD have native NFSv4 ACLs. They do however lack kernel oplock support so there are perhaps some caveats in that regard. On Thu, Jul 2, 2020 at 3:07 PM Strahil Nikolov via samba < samba at lists.samba.org> wrote: > Hi Kraus, > > I know that Gluster can be exported over NFS-Ganesha (supports v4.X), > Samba (protocol 1.0 in order to get 'real'
2007 Feb 21
2
How does Asterisk use SIP info command
What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu
2010 May 28
1
libsmbclient licensing
Dear Samba team, We have developed cross-platform multiprotocol intranet file searcher and it includes the module (SMB scanner for *nix) which uses libsmbclient to enumerate all files on smb shares ("uses" means including headers and linking with library). Other modules also use some external libraries, but all other libraries have LGPL license. W...
2008 Apr 22
2
Asterisk sends 486 Busy Here instead of 600 Busy Everywhere
Hi, We have a scenario wherein the endpoint needs to send a 600 Busy Everywhere after receiving an INVITE. I am using SIPp as this end point. SIPp is configured as UE2. Now when UE1 calls UE2 (SIPp) receives the INVITE and responds with a 600 Busy Everywhere. But when Asterisk receives this 600 response it sends out a 486 Busy Here to UE1. Ideally Asterisk should be relaying the 600
2005 May 26
1
SIP V2 Support
Dear All, I am totally new in this arena and I am still waiting for my installation process on freebsd to finish, but I wanted to make sure of the following: - Call routing between IP telephones can be done regardless of who made the phones? - Asterisk does support SIP V2? - it does act as SIP Proxy and Register? -- Thx MAG -------------- next part -------------- An HTML attachment was
2005 Jul 21
1
SIP & messengers & video phones
Is there a possibility to send text based messages from/to a sip phone (text display) or to a video phone or from/to a messenger? bye Ronald
2007 Feb 14
0
Zoiper softphone version 1.03 now available
...etter look and more advanced features. Finally MS Vista fans can also make use of it. Zoiper BIZ BETA is available free of charge from www.zoiper.com. There you can find out more about the improvements and features. We are also offering customization packages for ZoIPeR Free Windows. Zoiper is a multiprotocol: SIP and IAX / IAX2 softphone, supporting native conferencing, g729(optionally), Call recording, callto URL protocol, autoanswer and much more. For more information please consult: www.zoiper.com www.attractel.com Greetings, Mira
2007 Feb 23
2
Dial() command h and H options for SIP channel
Hi, Just need to confirm whether dial() command provided options h: Allow the callee to hang up by dialing * H: Allow the caller to hang up by dialing * work for SIP channels as well ? -ag
2007 Sep 19
2
what is softswitch
Dear all what is softswitch what is difference between asterisk and softswitch ?? regards satish patel --------------------------------- Catch up on fall's hot new shows on Yahoo! TV. Watch previews, get listings, and more! -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 May 17
0
Astricon 2004 - the developer's meeting ** CALL FOR PAPERS
...H.323 architecture * Configuration architectures o Dynamic and static data - how to separate o res_config and others * A common Authentication architecture * Extension and peer/user/friend configs * AGI and alternatives * Manager API development * Multiprotocol presence and messaging architecture for Asterisk * Documentation - the Wiki and the Doc project * Sexy things done with Asterisk - Bluetooth presence and others (showcase, including Mark's Wifiaxy ) * Voice: Vxml, Speech recognition and TTS modules * Instant messaging...
2005 Mar 03
3
Asterisk not relaying back the SIP response messages
HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling
1998 Sep 23
1
Project Cascade?
Hi all. I was wondering if anyone has heard of the new Sun product called Project Cascade? You can find more information about it at a link directly off the main sun.com page. I'm curious as to whether or not it will be as good as samba, and if people are going to think that an equivilent hasn't been around for quite some time... Dave
2015 Jun 15
3
Calling multiple phones at ones
...even see that people have written and submitted patches for this in the past, but they have been rejected: > > https://issues.asterisk.org/jira/browse/ASTERISK-13614 > > It has apparently been a somewhat contentious issue. Asterisk's philosophy is that it is not a SIP proxy, but a multiprotocol PBX that also happens to support SIP endpoints, and so the channel drivers need to be as generic as possible and anything that can be passed on to the dialplan to be handled in a uniform and consistent fashion should be, and that would include call forking. The developers do not want for there to...
2015 Jun 15
5
Calling multiple phones at ones
Hello group! I?m new to Asterisk but got one running finally :) Now I?m trying to solve following problem. I have company Automated Attendant and each employee have SIP phone at home, SIP phone in office, cell phone. I want all those 3 phones to be ?one?. So, if someone calls our company number and dials my extension - I?d like 3 phones to ring at the same time. What is this feature and where
1999 Jun 17
6
Samba vs. NetAppliance
Hi, I'm debating purchasing a NetAppliance fileserver that does native CIFS. Below is a URL to a NetAppliance authored paper regarding performance. One of the sections compares NFS to CIFS and talks about Samba. Can anyone dispute any of this? Is there any reason besides price that I should stick with Samba? -Ed Ed Sanborn (978) 691-6496 Northchurch
2015 Jun 15
0
Calling multiple phones at ones
...ituation. You can even see that people have written and submitted patches for this in the past, but they have been rejected: https://issues.asterisk.org/jira/browse/ASTERISK-13614 It has apparently been a somewhat contentious issue. Asterisk's philosophy is that it is not a SIP proxy, but a multiprotocol PBX that also happens to support SIP endpoints, and so the channel drivers need to be as generic as possible and anything that can be passed on to the dialplan to be handled in a uniform and consistent fashion should be, and that would include call forking. The developers do not want for there to...
2004 Aug 15
5
New $89 VOIP phone
Has anyone tried the new ariavoice $89 VOIP desk phone with Asterisk? ` http://www.voip-info.org/wiki-AriaVoice -- Jim James H. Thompson jht@lj.net
2015 Jun 15
0
Calling multiple phones at ones
...e have written and submitted patches for this in the past, but they have been rejected: > > > > https://issues.asterisk.org/jira/browse/ASTERISK-13614 > > > > It has apparently been a somewhat contentious issue. Asterisk's philosophy is that it is not a SIP proxy, but a multiprotocol PBX that also happens to support SIP endpoints, and so the channel drivers need to be as generic as possible and anything that can be passed on to the dialplan to be handled in a uniform and consistent fashion should be, and that would include call forking. The developers do not want for there to...