similar to: Samba Authentication - User ID Pass-Thru?

I know, why IIS when I can Apache? I'm actually running both... Kind of a hybrid network. Anyway, I'm trying to configure IIS to use a Samba share as its www root and I think I'm running into a security issue. See, in IIS, I have to connect as a specific user when I attach a network share as the www root. The share, from Samba's point of view, is read-only and every file in it

Hi, I have softphone with a g723 codec, my question is how do i set it as Pass thru in Asterisk? cheers, Aby Azid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080424/b442d5af/attachment.htm

What happens if asterisk receives a T.38 call? Will asterisk pass it thru? I've seen a few ATA devices that support T.38 and I'm wondering what happens if a fax is sent thru one of these ATAs into asterisk. Maby I have the terminology wrong. Is T.38 a protocol like SIP or is T.38 a compression like G729 using SIP? Thanks, Matthew

I've configured asterisk for pass-thru mode according the following two URLS: http://voip-info.org/wiki-Asterisk+G.729+pass-thru http://lists.digium.com/pipermail/asterisk-users/2004-March/039663.html I don't believe I have it working, as show channels reports that my calls are bridged. I'm assuming that bridged is the opposite of passed-thru...is this correct?

Hello, Trying to use g729 in pass-thru mode. Call flow: SIP IP Phone (G.729)-->Asterisk(1.6.2.9)--->SIP Trunk to ITSP(G.729) When using G.729, call is not getting connected. Below is the extract from CLI. == Using SIP RTP CoS mark 5 -- Executing [12127773456 at default:1] AGI("SIP/100-00000000", "call.php") in new stack -- Launched AGI Script

Hi all, Below is what I did to run Asterisk in pass-thru mode: sip.conf: [general] disallow=all allow=ulaw canreinvite=yes For each channel, canreinvite=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension

Hello, As I understand * don't supports T.38 in Zap channels (please correct me if I'm wrong, BTW is there plans for such support?) I believe it's should support T.38 in "pass-thru" mode. I mean setup like this: Hardware gate with T.38 ------ Asterisk ------ Hardware gate with T.38 But I had troubles with this setup (no faxing) while two gates conneted directly with same

Hello, I'd like to use g729 pass-thru when I dial out to a sip provider from my IP phone but because I have no license for g729 I'd like to use g711 ulaw for asterisk voicemail, conference bridge and other services. When I set in [general] section of sip.conf the following: disalow=all allow=g729 allow=ulaw the g279 pass-thru works fine with my SIP provider but when I call the

Hello, I have just updated my Asterisk installation from 1.2x to 1.4 (on FreeBSD) - mostly everything seem to work fine. However, I use G.729 pass-thru - and I have before successfully used the following setup: http://www.voip-info.org/wiki/index.php?page=Asterisk%20G.729%20pass-thru However, it is not working with 1.4 - I see the following errors: [Apr 16 15:59:24] WARNING[10139]:

Hello I am having a weird issue with a PCI-X Intel dual gigabit ethernet nic. While I am partially successful in pass thru with kvm, I have a weird issue with my pfsense freebsd vm. It see the two nics but gives them the same Mac address. I'm not sure if this is a pfsense bug or what. The nic works for everything but this Mac address issue. Long story short, I guess what I'm asking is

Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? We've tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback

Hi all, Here is my configuration: Phone ?? Asterisk ?? Gateway (SIP 2 PSTN) In the Gateway (patton) I have in "codec order" G729 then G711 If the Phone use G729, I have a pass-thru in the Asterisk ... It's the main case. But If I put G711 in the Phone, I want that the Asterisk try a G711 codec with the Gateway !! because in this case, it's G711 between the Phone and the

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/1572210f/attachment.htm

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/6386cacd/attachment.htm

Hi all, I'd like to know how Asterisk process a RTP data flow. Is there any clue to find out about this? The rtp.c? Thanks. Regards, Sam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050928/beb436a4/attachment.htm

Is it possible to pass PPTP packets through 2 firewalls before they hit the remote access server? I installed a Netgear ProSafe VPN firewall as the first line of defense in my network. I have since set up a Fedora Core 2 server running Shorewall 2.1.3 and Squid in non-transparent mode, between the Netgear unit and my network. So, the Netgear faces the Internet with a public, static, IP address.

https://bugs.freedesktop.org/show_bug.cgi?id=62305 Priority: medium Bug ID: 62305 Assignee: nouveau at lists.freedesktop.org Summary: No HDMI Audio out (pass-thru) on Geforce 7600 Go QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: doug at

I want to make sure that RTP is not going thru my asterisk. I read you should avoid in the dial commands: m music while ringing t,T transfer calls from caller and called party What else do I need to take care? remote phone ===> registered to local asterisk ===> calling remote gateway should have the RTP remote phone ===(RTP)==> calling remote gateway bye Ronald

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi All, This might be a stupid question, but I have two PCI devices that I want to pass-thru to a PV domU. I can pass one of them, but not both. I''m sure it''s a stupid syntax issue within my configuration file. When I do a "lspci" in the domU, (CentOS 5.1), only one of the devices shows up. (e.g.) 00:00.0 Multimedia video controller: Internext Compression Inc