similar to: Call redirect with *

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

Asterisk 1.2.7 RedHat 9.0 Hi, I've run into some voice degradation problems with IAX2: I frequently have calls come in on a DiD provided by an ITSP. I often have to redirect these calls back out to the PSTN (i.e. to a cell phone). When this happens, I don't want my server in the media path, I want to hand it off to my ITSP instead and let them handle both ends of the call. I've

Hi all, I have a group of public IPs 80.x.x.209-80.x.x.215. My satellite broadband router has .209 and my Fedora Core 2 running Shorewall firewall has one network card with IP .210 and the other with 172.16.0.1/255.255.0.0. All network PCs have a 172.16.x.x private address. I am new to shorewall and fedora/linux. I would like requests to public IP 80.x.x.211 to be redirected to internal IP

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.

Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time period. Then, if the call matches the second time period it need to be transfered to the trunk that directs to the second server. How do I do to configure it this way?

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=472 Summary: REDIRECT target translates all packets' destination to main IP of interface Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: RedHat Linux Status: NEW Severity: normal Priority: P2

Dear Sirs, I've got a weird problem with IAX2 transfers. My setup consist of 3 Asterisk servers. One is located in Europe on a public IP and a local PSTN connection through ISDN. Two are located in South-east Asia - both on public, but dynamic IP. These two each have a bunch of SIP phones attached. All 3 systems are running latest CVS (ok - might be a day or two old) on Linux 2.6.6 with

Dear Xapian community, Thank you to the developers and community for creating a supporting such a great search engine library! I wrote a tutorial on building a search engine using Xapian and Pylons (link below) and am featuring Xapian in a workshop this Tuesday, November 17 from 6:30-9:30pm on 23rd street and 6th avenue. Do you know people who might be interested? Please let them know! It will

Hello, http://www.tinc-vpn.org/examples/redirect-gateway/ explains how to set up a default route using tinc. However, in my understanding tinc's p2p traffic is routed via tinc there: Assume a network: Client A, Client B and a Server S. A and B have "ConnectTo S" in their config, S is used as a gateway providing 0.0.0.0/0 . All packets send from A to B will be delivered to

Hi, Another one. I got notransfer=yes i iax.conf for both 2109 and dialout, but I still get this in my logfile Attempting native bridge of IAX2[2109@2109]/5 and IAX2[dialout]/6 Asterisk Version is CVS-04/19/04-22:17:41 What's wrong ? I gues it has somethnig to do withe my bilsec-problem as well. /HHA

There are three optional ways to watch satellite tv on your pc (http://www.buydvb.net/blog/how-to-watch-satellite-tv-channels-on-computer.html), we?ll explore on this page, you can choose the most suitable way for yourself. Option 1: Connect a satellite dish directly to your PC Option 2: Connect your satellite box to a PC Option 3: Watch satellite TV channels over the Internet Option 1: Direct

I have a situation where our VOIP provider is running *, my office is running *, and my house is running *. I have an extension at the office so that if a call comes in from the VOIP provider and they select that extension, the call will be sent to my home * box and ring my phone. That works fine. I set "notransfer=no" in the iax.conf file at the office so that the office system can

On 4/20/2018 8:53 AM, Olaf Hopp wrote: > On 04/20/2018 02:01 PM, Olaf Hopp wrote: >> Hi (Stephan?), >> is it a new feature of dovecot 2.3 /pigeonhole 0.5 that a sieve "redirect" changes the envelope sender of >> a redirected mail or simply a bug ? >> >> A sends mail to B, B redirects to C >> C sees B (not A!) as envelope sender. >> It is not a

Trying to understand "redirect-gateway" option Is this url info correct for the current Tinc 1.0.35 version? .../tinc-vpn.org/examples/redirect-gateway What is the script? /etc/tinc/myvpn/hosts/server-up Is it really a extra bit to be added to,/etc/tinc/myvpn/tinc-up If I read and apply the instructions to my routed setup, then all I need to add isip route add 0.0.0.0/1 dev $INTERFACE