similar to: user list archive

On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: > Paul Albrecht wrote: >> Really? Shouldn?t something this major affecting the entire Asterisk >> community get discussed on the lists? Any idea what Leif is talking >> about when he says the community is in transition, moving from dial >> plan model to external control. > > It was

Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul Albrecht

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 I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

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

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 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

hello, I' trying to use Asterisk 12 Alpha. Compilation and instalation without issues. When I try to start asterisk with: asterisk -cvvvvvvvvvvvvvvv i see this error on the console: 17:09:43.559 sip_endpoint.c !Module "mod-refer" registered asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. Any hints? Thank you

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

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 =======================================

Hello! New message, please read <http://grillonwheelsnyc.com/told.php?65hg8> brettlist at nemeroff.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151022/5ac65990/attachment.html>

Hi, Many SIP phones implement list-based Notify-Subscribe mechanism with the phone may request to be notified of status changes from a whole list of resources. Thanks to PJSIP inclusion in Asterisk 12, I'm wondering how a Resource List Server could be implemented with Asterisk 12. 1. I couldn't see RFC4662 itself is implemented in PJSIP. Is this correct ? 2. Which architecture

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > 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

On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: > Same problem with me. > > I downloaded the file in 2 different places and had the same error... An issue was filed for tracking this[1] and it will be resolved later today. [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL

The 13.14 tar gz doesn?t even exists on the current or in the old releases folder. there seems to be an issue with the latest build not generating the artifacts? best regards On Feb 14, 2017, 11:04 -0300, Marcelo Terres <mhterres at gmail.com>, wrote: > Thanks Joshua. > Marcelo H. Terres <mhterres at gmail.com > IM: mhterres at jabber.mundoopensource.com.br >

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)

Marek ?ervenka wrote: > and what about > https://www.asterisk-blog.com/2016/02/17/odbc_gutting/ While not in the email these are listed in the CHANGES and UPGRADE.txt file. Going forward we'll try to ensure we include such things in the release notes as well. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:

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