similar to: Asterisk no audio on calls problem.

Hi, Thanks for your response on the kernel switching.I was away and could not reply immediately. Right now, I am facing a differentissue. I have to set up DNS server using BIND on Centos 4.3. When Itype the hostname on Centos, it shows: sipserver.vodcalocal.com But the cli prompt has root at sipserver~ meaning only the sipserver part of the hostname is displayed. whyis this so? What is the

I just tried to set up a tinc test installation, but I guess I did something wrong, because I can only ping one direction. Here's my setup (lan1,192.168.99.0)<-->firewall1<-vpn->firewall2<-->(lan2,192.168.100.0) firewall1 has tap0 on 192.168.88.2 firewall2 has tap0 on 192.168.88.3 i set up the following routes : on firewall1 : 192.168.100.0 gw firewall2(192.168.88.3) on

I have 2 firewall in HA with keepalived. Can I use active the same tinc configuration on 2 firewalls ? using tun Interface with same ip on all 2 nodes is a problem ? tun device advertise itself on the network having an IP/MAC pairs (ARP) or the IP is only used by the system internally for routing so using the same configuration is right ? so one firewall be active, the other is passive. With this

I think it should work at least for TUN virtual interface as TUn works at IP level. This is a sample configuration. firewall1 lan = 172.16.1.11/19 (ALWAYS ACTIVE) - "Physical Network Interface" - system config as ifcfg-... 172.16.1.10/19 (VIP Keepalived Make active) - Active/Passive configuration with firewall2 firewall1 vpndr1

Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register =>

Hi. I've just bought SIP telephony service from a Swedish telco. I've managed to make and receive calls with kphone. Now I want to set up asterisk to be able to add fancy features like voice mail and recording conversations. But first I have to get the basic setup right. I'm running asterisk and kphone on the same machine, behind at NAT-router. When I make a call (from my regular

This is a vpn for Disater Recovery sites, so it is not necessary to have a seamless failover, strictly speaking. Encryption instead is mandatory. Testing we found that on Keepalived failover remote Tinc take few seconds to reset the connection and correctly re-connect to the new active firewall (probably new firewall resetting the connection + PingTimeout + some seconds to reconnect). This is

This is what I want to avoid :D I want an active Tinc virtual interface active with ip identical of the other firewall, without ip conflict on the same network. Do you know if Tun type virtual interface on one host can have same ip address of another host in the same network without ip conflict ? ie if a tun virtual interface can work active without transmitting on real network ? or if such a

Hi Saverio, I found conflict: 172.16.1.10 00:50:56:1b:ba:5e VMware, Inc. 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 2) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 3) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 4) 172.16.1.10 00:50:56:2b:12:e6 VMware, Inc. (DUP: 5) So my assumptions were wrong ! :D Probably Virtual

Hi, Is there a specific latency that asterisk accepts? I encountered a problem wherein when the latency was unusually high,my xlite's (i have 2 xlite) cannot register. but when the link suddenly went stable, the x-lite just registered. what i forgot to look at is if the registration packet is reaching my asterisks. ------ when xlite cannot register --------------- Pinging

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

Kevin wrote: > There is a reboot script posted on the wiki to reboot Polycom > telephones. When I execute this script, I get the following messages. > I am concerned as this is causing issues with asterisk and the PRI. > Does anyone have any ideas why this would be happening? > > > > asterisk console: > > -- Remote UNIX connection > -- Remote UNIX

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

With my thanks to Brian West and his offering in the thread, "Subject: Re: [Asterisk-Users] Remote reload Cisco 7960" I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones PolyReboot.pl takes an IP address as a single argument and reboots the phone. You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all lower case. Further, you need to use ftp for

Why is SER considered a better SIPserver than asterisk , why is it that SER can handle more clients than asterisk can. And if this is just cause of say poor SIP handling code in asterisk then is there anything being done to fix it. Just wanted to know why SER claims to be better than asterisk as a SIP server. ? -- regards Vikram (http://www.vicramresearch.com)

Hi All, I have a thin cliente here that i want to run asterisk: - National Semicondudor Geode GX1 266MHz Geode 266MHz single chip - NS Cx5530a Southbridge National Semiconductors SC2200 - NS PC97317 in chipset - 32MB Compact Flash - 64MB Ram - 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816 Some tip? I have a ide>flash

Hi, I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications. You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example) ? /**************************/ simple HTML code example: /*************************/ <html> <head>

Hi all! I get this error when trying to start asterisk: ERROR[8192]: File asterisk.c, Line 1349 (main): Unable to connect to remote asterisk What can be the problem? Thank you! Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 UK +44 870 - 3403539 FWD 64662 sip:ipfone@sipserver.com.br www.ipfone.com.br info@ipfone.com.br -------------- next part

Hi list, I try to install the following network: /-----------------\ | Firewall 1 | \-----------------/ ___________________| |_________________ | 10.2.6.1 10.2.2.1 | | eth1 | | 10.2.6.11

Samba version 3.0.9 on Red Hat 9.0 workstations: NT 4.0 SP6 When I took over this project, there were two domains connected by a VPN. For reasons that I'm unable to understand, some new users wound up with the intended domain of SATA and some wound up in SATB (even though they joined the SATA domain. Recently, all kinds of problems have materialized and the only thing I can find wrong is