similar to: Asterisk and IP Aliases

Hello, Hope somebody can help me - Asterisk is behaving very oddly and I'm totally stumped! I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a

Hello, I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf

Hi all I am in this strange situation: we had ser configured to relay calls to numbers to asterisk extensions and all used to work nicely, with both ser and asterisk running on the same machine with public ip (ser on port 5060 and * on 5061). We had to move temporarily our server to another provider which put our server on a dmz, so that now we have our server with private ip but reachable from

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

I have an * box that I need to chang the IP address on. My hope was that I could add an alias to the interface with a different IP address, have * bind to all addresses, change DNS and when no more hits come on the old address. However, IAX registrations coming in to the alias don't seem to get acknowledged by *. Even with iax2 debug on, I don't see any attempts. We can ssh in on both

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified

Hello, AFAIK, aliases cannot be set for Graphics devices. I tried this very simple experiment with libvirt 1.2.4 (from virt-preview on F20) relevant part of the input xml: <graphics type='spice' port='-1' autoport='yes' listen='127.0.0.1'> <listen type='address' address='127.0.0.1'/> <alias

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

Well, this is a major pain in the ass. I got realtime static working for sip.conf. 'Great!' I thought. That was until I realised I couldn't use it. Our Asterisk systems are using OSPF and listen on interface lo:1. Asterisk doesn't like to use an interface name for it's bindaddr setting, so you have to put the IP address of lo:1 in there. If you put in 0.0.0.0, it seems to

Valued Colleagues, I am trying to configure and use asterisk manager API. The /etc/asterisk/manager.conf and the output of "netstat -nl" are appended below. When I restart asterisk, I believe I should be able to see the asterisk listening on port 5038 using netstat. But when I type netstat, I don't see any applications listening on port 5038. When I telnet to port

On Mon, Feb 23, 2015 at 5:51 AM, Raj Roy Ghandhi <roy.gandhi at gmail.com> wrote: > Hi Friends, > I encountered a strange issue. > I am running Asterisk 11.8.1 on Cent OS with no firewall running. > It has 3 NIC interfaces. > > in my sip.conf I have > > allowguest=yes > bindaddr=0.0.0.0 > udpbindaddr = 0.0.0.0 > > But my Asterisk instance does not pick

Hi folks, Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

----- Original Message ----- > From: "Daniel P. Berrange" <berrange@redhat.com> > To: "Eric Blake" <eblake@redhat.com> > Cc: "Francesco Romani" <fromani@redhat.com>, libvirt-users@redhat.com > Sent: Wednesday, May 14, 2014 7:24:13 PM > Subject: Re: [libvirt-users] aliases for graphic devices > > On Wed, May 14, 2014 at

On Wed, May 14, 2014 at 10:28:13AM -0600, Eric Blake wrote: > On 05/14/2014 09:47 AM, Francesco Romani wrote: > > Hello, > > > > AFAIK, aliases cannot be set for Graphics devices. I tried this very simple experiment > > with libvirt 1.2.4 (from virt-preview on F20) > > > > relevant part of the input xml: > > > > <graphics

On 05/14/2014 09:47 AM, Francesco Romani wrote: > Hello, > > AFAIK, aliases cannot be set for Graphics devices. I tried this very simple experiment > with libvirt 1.2.4 (from virt-preview on F20) > > relevant part of the input xml: > > <graphics type='spice' port='-1' autoport='yes' listen='127.0.0.1'> > <listen

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

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be

Hi Friends, I encountered a strange issue. I am running Asterisk 11.8.1 on Cent OS with no firewall running. It has 3 NIC interfaces. in my sip.conf I have allowguest=yes bindaddr=0.0.0.0 udpbindaddr = 0.0.0.0 But my Asterisk instance does not pick the call at all. When I check the listening apps using lsof -i I get asterisk 3046 asterisk 7u IPv4 1191172 0t0 TCP *:5038 (LISTEN)