similar to: Native architecture never available in menuselect

Has there been a change in the way certified Asterisk is being packaged? Starting with certified Asterisk 11.6 has all the extended options are checked by default in menuslect? Certified Asterisk 11.2 does not have them checked and neither does certified Asterisk 1.8.15? Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I did what we use to dim that is add a line to pjsip.conf like #include /etc/asterisk/pjpeers.conf but the file is not loaded. Am I doing something wrong this functionality is disabled?

I cannot wait for the regular bug-patch process to play out. I am considering hiring a developer to fix bug 24015, and of course submit the patch for the bug. The issue is simple, the app Transfer does not transfer when using PJSIP.. I called Digium and they said that they do not do this kind of work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 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

?Asterisk 13.11 rc1 ./configure LDFLAGS="-z muldefs" --libdir=/usr/lib64 --with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode --with-pjproject-bundled ?checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no checking for pjsip_tsx_create_uac2 in -lpjsip... no checking if "pjmedia_mod_offer_flag flag = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using

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

?I need to install Asterisk on a MAC, native, no virtualization. Has anybody done this? Are there documents on the Internet? I googled it and all web sites that claimed to help installing Asterisk on a MAC have disappeared. Is it possible at all? Digium should actually have a MAC app in the Apple store with a PBX. It should be a paid app. I would buy it right away. ? -------------- next part

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

Hi, Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such a way that I cannot script non-english sound files downloading anymore. The following used to work (unattended) with 1.6.1.9 (for instance): cd /usr/src/asterisk-${ASTERISK_VERSION} ./configure make menuselect.makeopts echo "MENUSELECT_CORE_SOUNDS=CORE-SOUNDS-EN-GSM CORE-SOUNDS-FR-GSM" >

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>

Hello List, Is it possible to select ODBC_STORAGE without entering to 'menuselect'? I'm currently building a package for my distro with a little script, and would like to set this option without entering manually to 'menuselect' I know that I could make the script to change the 'menuselect.makeopts' var from: MENUSELECT_OPTS_app_voicemail= to:

Hi All, I'm in the process of writing an install script and I would like to change some settings for the install process but I don't want the user to go into menuselect and make the changes manually. Is there a way to make the changes to menuselect from the CLI? As an example, selecting the iLBC codec. menuselect codec ilbc on Regards David. -------------- next part -------------- An

Hello List, I've been trying to compile Asterisk with H.323 support and, after correctly installing PTLib and H323plus (OpenH323), the Asterisk configure script still doesn't detect the dependencies as installed. I know they are correctly installed because after going into "[asterisk-source-directory]/channels/h323" and issuing a 'make opt', it correctly builds

I'd like to be able to save the choices made in menuselect in a way that they can be tracked in a CM system and applied to a later release of Asterisk using an automated tool like Ansible. What's the best way to do that?