similar to: Getting better Caller ID

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

Hello, I have a question regarding incoming fax to local file (on the Asterisk server). While the fax is received properly (I have the tiff file generated as expected) I get the warning 'FAX CNG detected but no fax extension' on the consol. If the fax is received ok then what 'fax extension' does it expect and what should I do there? My Setup: Sender -> Public PSTN ->

Il 20/04/2017 17:32, kevin.larsen at pioneerballoon.com ha scritto: > > This gets kinda Rube Golberg-ish, but convert the incoming analog line > to sip, route it through asterisk and have asterisk do its thing > before converting it back to analog to send to the phone. Only problem > is you get a lot of extra hardware involved in the mix to make it > work. It will be a lot of

True agree, problem is somehow the people purchased.... am supporting to overcome that. Trying level best... around 20 phones has been purchased.... ?? On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, <mefhigoseth at gmail.com> wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bought a Grandstream IP phone or Yealink... > -- >

Hello list, I have Asterisk running well inside our network. I did some experiments exposing it to internet but had some issues: 1. NAT issues (voice one way, etc). From what I understand double-NAT users will always have something like this 2. Immediately I see people trying to hack into. I did configure Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc So.. I ended up closing

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten =>

Steve Edwards <asterisk.org at sedwards.com> schrieb: > You can fiddle with the ring tone by phone specific configuration and > phone specific SIP headers (sipaddheader(Alert-Info: ...)). > > These seem relevant: > > http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the > discussion looks relevant as well). > >

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Make sure you have solved the problem. You don't want to get hit with a > phone bill for calls from your location to Israel. Basically, they are > hoping that you are running the equivalent of a mail server open relay. > They are trying to use you to dial out to another number. You don't want > to pay

Hi, I've some analogic lines and I'm asked if it's possible to program an asterisk for "checking" the inbound calls without answering them, doing something like this: analog line 1 -----+---------- asterisk | \______ analog phone when a call enter, asterisk sense it and store its values (callerid, date and time, etc) somewhere, but

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

2015-06-08 22:35 GMT+02:00 D'Arcy J.M. Cain <darcy at vex.net>: > On Mon, 8 Jun 2015 22:24:33 +0200 > Luca Bertoncello <lucabert at lucabert.de> wrote: > > Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > > > Basically, they are hoping that you are running the equivalent of a > > > mail server open relay. They are trying to use you

Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as "maxBitRate". Without capital "M". Are you closer to

We have following problem. On some of the extentions I call cell phone after 10 seconds or so.Or, like this one below- we call cell and office phone at the same time ;Eric on extension 105 exten => 105,1,Dial(${ERIC_CELL}&${ERIC_OFFICE},30)         same => n,VoiceMail(105 at default,u) Where problem comes in - if person not at the desk - his cell phone shows call from OFFICE number and

Thanks for your response. I am able to capture a few more lines of the dump (see console-output.txt). dmesg output attached. My bitmap is 720x512. Below is the library versions, and amount of vram. Any feedback is appreciated. Thanks. Steve [steve at localhost ~]$ lspci -nn -d 10de: 01:00.0 VGA compatible controller [0300]: NVIDIA Corporation GF108 [GeForce GT 430] [10de:0de1] (rev a1) 01:00.1

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone

Hi, Would the Windows 2008/2012 server be looking for a particular DNS record during DCPROMO? Both the Samba and Windows box are on the same vlan/host/subnet. The UFW has been disabled. Stupid question, do I need to install any RPC package in the Samba box? Would disabling Bind9 using dnsupdate and dns in server roles help? The only issue I see with that is the SRV records will disappear and

Hi Rowland, I did that initially and that came with Failed to connect to ldap URL 'ldap://lin-pdc.lin - LDAP client internal error: NT_STATUS_BAD_NETWORK_NAME Hence I removed the whole ldap:// bit After your email I tried again but using ldap://localhost and it seems to have worked. Not sure what the issue is with the fqdn. I could run ldap queries when using fqdn. Regards, Praveen

Hi All, I am using RHEL 6.4 on a Dell Server with 32 cores. But in the Cpus_allowed_list only 30 cores are available. Why is that? See the snippet from cat /proc/self/status: Cpus_allowed: 7fff7fff Cpus_allowed_list: 0-14,16-30 The allowed list is same for the init process as well (see below). I didn't see this issue, in RHEL 6.2. Why is that? Is there a way to change it? -miraj