similar to: 1.6.2 : global vars not read/set after #include w/ globals

Hello. I would like to configure Asterisk to accept incoming calls from two different GoogleVoice numbers via gtalk and jabber. I'm running Asterisk 1.8.3.2 and I can get one number working just fine. However, I can't figure out how to modify the gtalk.conf file shown on the Asterisk wiki site to work with two different jabber profiles. Do all incoming GoogleVoice calls have to go

I'm trying to allow access to the office from home. But the ip provider (cablevision) blocks udp 5060. I can see the register packets leaving on wireshark, but nothing received by office. Changed to port to 6111 and now the packets show up. In the server I've set port=6111 in the device in sip.conf, but * is NOT listening for 6111: netstat -an | grep 5060 tcp 0 0

I just wanted to send out some information that will hopefully help others. I don't know, maybe I'm the only one that's been having problems with this. I've been pulling my hair out for a while wondering why Google would not send my incoming calls to my Asterisk box. The calls would just roll to voice mail and no packets ever reached Asterisk. This has happened on two separate

I've been lurking on the dev discussion on creating nat=auto. It all leads me to think there's no reason to use nat=no. We have about 60 internal sip extensions connected to an multihomed asterisk box where the external ip is not nat'ed. Each of the internal sip contexts has nat=no. On startup I get a slew of warnings about intruders being able to distinguish real extensions. But

I can't get a test T.38 fax between 2 1.6.2 machines, using app _fax and spandsp pre17 and 20100501. The machines can't seem to get connected. send side extensions.conf: [fax-tx-test] exten=>s,1,NoOp(Context fax-tx-test) exten=>s,n,SendFAX(${FaxFile}.tif) exten=>s,n,HangUp() exten=>h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} FAXMODE: ${FAXMODE})

I'm currently running Asterisk 11.2.1 and I've noticed that when asterisk has been up for a while (usually about a day), outgoing calls through GoogleVoice fail to complete. I hear it ringing on my end but the caller never hears the phone ring. A simple restart of Asterisk seems to clear it up for another day or so. Has anyone else noticed this? -- Chris -------------- next part

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

Trying out 10.0.0-rc1. It dies starting up: == Parsing '/etc/asterisk/codecs.conf': == Found [Nov 11 17:07:05] WARNING[5078]: translate.c:1060 __ast_register_translator: empty buf size, you need to supply one [root at asterisk ~]# Where do I supply the "buf size" to the translator? And what should it be?? sean

On 12/02/2014 02:46 PM, Jeffrey Ollie wrote: > On Tue, Dec 2, 2014 at 1:22 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >> Or do I >> find a new place to put asterisk.pid? > > Also, if you use the native systemd unit file, you no longer need a > PID file, although you still need /run/asterisk to store the control > socket. > So systemd is taking

On Fedora 20, every time the kernel updates, /var/run/asterisk owner is set to root.root. I'm running asterisk under user asterisk. Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I find a new place to put asterisk.pid? sean

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear

android email will not play wav49 file attachments. See: http://code.google.com/p/android/issues/detail?id=1712 Now I'm getting a lot of pressure to change the format used in voicemail. Here's what I've got: format = wav49|gsm I'd like to change it to format = gsm|wav49, but the voicemail.conf.sample says "Don't Change the Format Unless You REALLY Know What

I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using

We have a 10.10.0.0 internal network. The asterisk server - 10.10.10.180 - has a PRI connection to a T-1. Another server is the router to the internet. All phones in the office and the workstations are on the network. Most of the internal phones are aastra 9133i's. Here the network config from a phone: Network Settings Basic Network Settings DHCP [ ]

I've got a confbridge set up which works if dialed locally: -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing

On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a

On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: > > https://www.voip-info.org/asterisk-security/ > > > > -----Original Message----- > From: asterisk-users [mailto:asterisk-users-bounces at

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM