similar to: Asterisk Proxy Type

Displaying 20 results from an estimated 800 matches similar to: "Asterisk Proxy Type"

2004 May 03
1
Asterisk & MGCP / NCS
Hi everybody, I have a MTA from Terayon that I try to make run with Asterisk using MGCP channel. The device is running with MGCP 1.0 NCS 1.0 Each time Asterisk try to send a Request (Request Notify, Audit Endpoint....) the device returns error 510 "Protocol Error" Does anybody have already meet this problem and provide me support to make run it ?! (I have already try to change
2004 Apr 26
2
Registering a Grandstream Budgetone with Asterisk from Home
Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know
2004 May 07
4
SIP Wokflow diagram
Hi everybody, I would like to create SIP call flow Diagram under Windows. Is anybody know a program to perform it? I have already Ethereal and I would like an explicit diagram just to show where something have problems... Thanks Ignace
2004 May 05
1
Early B3
Does anybody can explain me what is early B3? Thanks! Ignace
2004 Apr 23
1
CAPI and Extensions.conf Security problem
Hi, I've installing a AVM Fritz Card in my ASterisk Box I've configured everything and its running perfectly. The problem is that everybody is allow to call through it. Explaination: All users registered in Asterisk can make a call towards the ISDN network But, everybody from the Internet, knowing the extension of CAPI in the dialplan, can call through my Asterisk to any phone
2004 May 28
1
Immortal SIP & NAT problem
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
2004 Apr 29
1
Need an explanation about different protocols
Hello, Is there someody who can explain me the meaning of these sentence. "Sip is philosophically horizontal and H.323/MGCP are vertical" Thank U (if you have some links to share about this protocols, share it :) ) Ignace
2006 Oct 30
3
Grandstream ATA 286 tdm400 and Asterisk 1.2-13
Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme
2008 Feb 14
1
Ser, asterisk and ip2ipgw
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1">Hi,<br> <br> i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull
2004 Aug 15
2
GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)
Hello everybody, when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286 and H323 EP (my post from 13/08/04) I checked further and discover that problem is with ATA286 who is unable to call. I always get an 404 error. Coming back to RC1 everything works fine again. I tried to modify my dtmfmode from rfc2833 to info but in change nothing. Local call to asterisk are
2012 Jun 22
1
[PATCH] Fix wrong NFS umount path
Hi, When mounting a NFS share and an error occurs for some reason, the NFS share is not unmounted correctly: the local path is used instead of the remote path in the umount_v[23]() call. This patch fixes this. I found this while debugging some problems mounting the root file system on NFS in a test system; I saw this in the logs (/mnt/duplicated is the legitimate remote path): Jun 21
2011 Sep 15
1
Xrdp
The VNC server just listens to the appropriate TCP/IP port and then runs Xvnc which does the actual VNC communication. Ideally I'd be able to do the same thing for RDP then the daemon doesn't get any more complicated, and a bug in the RDP layer can't crash the server. I don't know enough about how NX works but I suspect we could do the same thing as for VNC and RDP. I'm
2020 Feb 27
9
[Bug 1410] New: STATELESS, rules with notrack into a map
https://bugzilla.netfilter.org/show_bug.cgi?id=1410 Bug ID: 1410 Summary: STATELESS, rules with notrack into a map Product: nftables Version: unspecified Hardware: x86_64 OS: Debian GNU/Linux Status: NEW Severity: enhancement Priority: P5 Component: nft Assignee: pablo at
2005 Jun 15
3
Grandstream ATA Toasted
A BETA firmware upgrade toasted my ATA286. It now has limited operations. It will get an IP address via DHCP and register to the last configured SIP server, but the web interface is gone as is the voice config menu. Apart from registration, there doesn't appear to be any other SIP functionality. An Ethereal dump does not show the device trying to grab a new firmware via tftp on bootup, so
2004 Dec 13
7
Dialing out to 2 clients simultaneously
Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same
2007 Nov 14
4
Hardware Requirements for qdisc htb/sfq
I am planning to replace our cisco 7200 core router with Linux. We currently serve around 1500 (3/4 DSL - different router) customers with probably half of them being concurrent at any given time. We have a fiber network and customers currently aren''t managed as far as how much bandwidth they can use at anytime. Therefore I have constructed a working tc qdisc Linux router as a test. It
2004 Jan 09
2
High speed traffic filtering
Hi; First, sorry if this question is mostly netfilter related, than lartc, but I think you guys may have a your opinion about this. I''m using Linux 2.4.x with netfilter packet filtering / NAT on our front-end firewalls (P500 with 1Gb RAM), which are filtering traffic going to our Public Web Sites. The traffic is growing very fast since several months.. The average traffic filtered by
2010 Mar 16
4
Get content from HTML element in Rails
Hey, I''m looking for a way to get the content of my div in my Rails view. I would like the literal HTML content, so only HTML and no Rails I thought inner_html would work, but I get a RJS exception (TypeError: $("content").innerHtml is not a function) Does anyone know how I can get the content? Thanks! In my controller I use this code: [code] def save_test render :update
2007 Nov 19
1
biocep project (R for the Web and the Virtual R Workbench)
Dear all, I have been writing during last year at the European Bioinformatics Institute a general unified open source solution for R integration. This work is now available via this link: http://www.ebi.ac.uk/microarray-srv/frontendapp/ The different frameworks and tools of the biocep project are now robust enough for production use. The different APIs are finalized but the documentation is
2005 Jun 03
2
Simple sip.conf question
How do I match by username instead of by host/ip? By default this is how it should work, but it does not. we do not have insecure turned on. Matt