similar to: important message

On Tue, Dec 23, 2014 at 6:34 PM, Rusty Newton <rnewton at digium.com> wrote: > On Tue, Dec 23, 2014 at 4:17 PM, Joseph <syscon780 at gmail.com> wrote: >> Are there any adapters that would allow me to connect asterisk to wifi or we >> are not there yet? >> I have Digium adapter S101i that was discontinued but similar device that >> would connect to wifi

Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server

<div>Is there any chance to create feature request for that useful functionality?</div><div>š</div><div>02.07.2015, 14:03, "Rusty Newton" <rnewton@digium.com>:</div><blockquote type="cite"><div><div><div>On Wed, Jul 1, 2015 at 4:46 AM, <span><<a href="mailto:royj@yandex.ru"

Hi, all Is there someway ability to insert custom Header to "SIP 486" message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment.

Are there any adapters that would allow me to connect asterisk to wifi or we are not there yet? I have Digium adapter S101i that was discontinued but similar device that would connect to wifi network and a cell phone would be handy. -- Joseph

Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I'm not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

<div>Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action.</div><div>Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affectš<span>performance.</span></div><div>š</div><div>02.07.2015, 15:31, "jg"

Hello List, In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots of below log entry (asterisk version 1.8.9.3). Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol chan_sip.c: Purely numeric

I have two contexts and there I have some sip clients and some iax clients, in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax clients I have some extentions like 2000, 2001, 2002, 2003, etc. My extention is 2003 when I make a call the manager program show me that the extention 20 was originating the call (in red, and show my name/extention)! When I make a call to

Hello - I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an "internal" eth0 and an "external" eth1. In pjsip.conf, I define the following transports: [trusted] type=transport protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use

On Thu, Apr 2, 2015 at 10:43 AM, Rusty Newton <rnewton at digium.com> wrote: > On Wed, Apr 1, 2015 at 9:08 AM, Trey Hilyard <kctrey at gmail.com> wrote: > >> Hello - >> >> I am trying to decide if I have stumbled across a bug in PJSIP or I am >> just missing something. My Asterisk has two interfaces, an "internal" eth0 >> and an

Hi All, I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? Basically I do a DB get/put before the call is placed. After the call is completed I want to do another get/put; however the dial application dies when the originator hangs up. Any way to get around

On Tue, Dec 23, 2014 at 6:51 PM, Joseph <syscon780 at gmail.com> wrote: > > > Most cell phone don't have a USB port but you are correct, maybe I just need > IAX2 soft-phone like: > Zoiper - it works on most of the platforms. I think Zoiper registers > directly with Asterisk IAX2 (if configured) as an extension, isn't it? If your cellphone is capable of a Wi-Fi

What CRM your going to use? With regards N.Prakash From: Rusty Newton Sent: ?28-?06-?2014 01:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Popup URL for outgoing calls. On Sat, Jun 21, 2014 at 5:57 AM, Inventions <research at businesstz.com> wrote: > Can anyone tell me how to implement a popup URL native asterisk when > making

hello, is it possible simultaneously use chan_sip and chan_pjsip? if yes, can you recommend settings i'm thinking about - chan_sip - for sip hardphones/softphones (sip udp 5060) - chan_pjsip - for webrtc -- --------------------------------------- Marek Cervenka =======================================

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)

Dear all, I have made a fairly complex dialplan where I am using the REGEX function in many places. This works so far, but I wasn't able to solve the following problem. What I would like to do is the following (please note that this is normal regex syntax and obviously not what the REGEX function expects, but I hope it shows the idea): same => n(A1), GotoIf($[${REGEX("^\+49.*"

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

Hi All, I had an unusual problem today; I'm sure it's a configuration problem. I had 2 phones behind a nat device and I had qualify=300 in both extensions config. The device I was talking to was an edgewater traffic shaper,/Sip Proxy. Since it is acting as a sip proxy, it was ignoring the OPTIONS messages that * was sending, and thus * interpreted that as the extensions being down. I