similar to: SIP Redirect from Asterisk behind a NAT

Hi all, I need some help, advice or whatever you can explain to me because I haven''t got a clear idea about how to do the following assembly, and I''d be very grateful if I got some help from an expert like you. I''m trying to build a system which represents the following: I''ve got a hosts unit (host1, host 2, ...) which have IP in the network 192

Hello, I have a dataframe (Lx) with 5 Lb, and 5 Lw variables. I want to create several variables according to the loop presented below (data attached). Lx <- read.csv("Lx.csv", header=T, sep=",") for (i in 1:20) { Lx$sb1[i] <- Lx$Lb1[i+1]/Lx$Lb1[i] Lx$sb2[i] <- Lx$Lb2[i+1]/Lx$Lb2[i] Lx$sb3[i] <- Lx$Lb3[i+1]/Lx$Lb3[i] Lx$sb4[i] <-

HI LIST CAN U HELP ME If there are multiple sip trunks with the same ITSP then an incoming call is arbitarily matched to the last peer with the same host IP address. This is not a serious problem because the DID is still correct but it does have many insidious effects due to the incorrect channel name Example register=myaccount1 at sip.myitsp.com/line1 register=myaccount2 at

If a call comes into my Asterisk server on a DiD provided by an ITSP and the dialplan sends that call to another external number throught the same ITSP's network, I don't want the RTP packets to pass through my server once the call is bridged. I have had great success getting this to work using IAX, but I have not been able to get this to work with SIP. The call is bridged OK (media at

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

Hi, I'm having trouble getting Dial to exit when the caller hangs up in Asterisk 1.4.21.2. I use a POTS line to call into the DiD given to me by VOIP service provider. When the call comes in, I have the VOIP provider send it to another POTS line. All this works fine however when the caller (me) hangs up, the Dial command does not exit. The callee stays connected (and my billing

Hi all: I get the following error message that I am not able to resolve. Error in if (const(t, min(1e-08, mean(t)/1e+06))) { : missing value where TRUE/FALSE needed It appears right before the last data.frame statement. Below is the program that simulates data from one way random effects model and then computes normality and bootstrap confidence interval for

I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request? # sip.conf [general] insecure=very permit=207.148.115.10/255.255.255.0 [myproxy] type=friend host=217.118.115.10 context=from-sip # Logging: <--- Reliably Transmitting (NAT) to 207.148.115.10:5060 ---> SIP/2.0 407 Proxy

Hi, I am in the process of migrating a system that was using KVM to VMWare Workstation. The original code needs to list all the domains available on the local machine, no matter what their state is. The problem is that the shut off domains are not shown by the virsh commands on the VMWare Workstation system. On the KVM system, with one shut off domain, I get this output: maxpower at sw ~$ virsh

I have been unable to understand the connection between an IAX registration for dynamic IP assignment and and the host definition. I have signed up with an ITSP for a DID. My ip is dynamic and although I have a dynamic DNS name, we are registering and outbound works fine. I'm at a loss to understand the relationship between the registration and the [section] definition in iax.conf that will

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Sorry for being sos insistent. Any helpful idea'?? ---------- Messaggio inoltrato ---------- Subject: [Samba] Problems with samba under FreeBSD, not under Linux Date: 19:02, mercoled? 24 novembre 2004 From: Vittorio <v.demartino2@virgilio.it> To: samba@lists.samba.org Dear All, (Context: Office windows LAN; PC Pentium 3 with 128 MB, FreeBSD 5.3.) Here you are the unanswered

I am having a problem with mapping drives to a samba server (1.9.18p4 and 1.9.17) that is very reproducible. The 1.9.18p4 is a modified server that has some special krb5 support in it (in addition to the stuff you previously added that I sent you). Here are two examples that are repeatable. Each is preceeded by killing any samba daemons that are for connections from the host in question.

I'm installing Samba 3.0.0 on a RH9 system for the first time. I've gone through the doc and I believe that I've done everything correctly, but when I try to issue a "net ads join -U myuserid" command, I get no response. No errors, just nothing. When I do a klist I can see the credentials for "myuserid" but nothing else. If I do a "net rpc join -U

NO AGENCIES PLEASE With that out the way, we (http://www.firebox.com) are looking for a web developer to join our existing small development team. Here''s the posting from our website[1]: ==== Web Developer (ref W200) Reporting to: Technical Director This permanent, full-time role involves developing the Firebox website and internal admin tools. You will be working in a small team on

Dear All, (Context: Office windows LAN; PC Pentium 3 with 128 MB, FreeBSD 5.3.) Here you are the unanswered message I sent to the FreeBSD mailing list: \BEGIN{MESSAGE} -------------------------------------------------------------------------------------- After installing and launching samba 3.0.7 daemons under my postgresql FBSD5.3 stable server at office, I'm having trouble in connecting

Hi, I have a quick question regarding estimation of a truncation regression model (truncated above at 1) using MLE in R. I will be most grateful to you if you can help me out. The model is linear and the relationship is "dhat = bhat0+Z*bhat+e", where dhat is the dependent variable >0 and upper truncated at 1; bhat0 is the intercept; Z is the independent variable and is a uniform

Hello Everyone, I am new with XCP. I''ve setup several hosts with* XCP 1.0* and manage it through XenCenter and command console. Each host has two NICs, one (xenbr0) connected to a physical switch for Internet (said sw1), the other (xenbr3) connected to a physical switch for internal network (said sw2). I am trying to setup VMs and VLANs on xenbr3. I created a virtual bridge by

Hello community, I've been running Asterisk on an embedded device for about six months, and my operation has been largely trouble-free. I'm hoping I could get some help with a minor problem: Every week or three, my PBX gets stuck in a state where it can receive calls, but it becomes completely unable to originate outgoing calls until I do a "sip reload". After doing the SIP

Hi, How is iax.cc / sixTel to be configured as a termination provider in asterisk@home? The iax.cc / sixTel instructions tell you to do this: iax.conf: ------------ [sixTel] type = friend host = iax2.sixtel.net context = inbound secret = mypassword allow = all extensions.conf: -------------------------------- exten =>