Re: [asterisk-users] rasberry pi

Im using two Rasp running debian with Asterisk 11 and 3 concurrent call with usb dongle as mobile trunk for each rasp and no issue until now :) On Wednesday, 6 July 2016, D'Arcy J.M. Cain wrote: > On Wed, 6 Jul 2016 01:10:23 -0700 > Thufir >

Re: [asterisk-users] rasberry pi

On Wed, 6 Jul 2016 01:10:23 -0700 Thufir wrote: > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a > SOHO, maybe three hardphones, rasberry pi would suffice? I would be > amazed, but, if so, great. I haven't used it extensively but I run Asterisk on an

Re: [asterisk-users] rasberry pi

I use RasPBX on RPi3. It is rock solid and feature rich! On 6 July 2016 at 11:51, Thufir wrote: > ok, that's really all I need to know. Of course, if anyone else wants to > throw in their two cents, don't let me stop you :) > > > -Thufir > > On Wed, Jul 6, 2016 at 1:36

Re: [asterisk-users] OpenSIPS or Kamailio based fronting for Asterisk?

On 5 July 2016 at 17:18, Tickling Contest wrote: > Hello, > > I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so > far my biggest issue is the complete lack of quick-start-like documentation > for either. Is there any place I can get a very

Re: [asterisk-users] rasberry pi

Another good choice for SOHO applications is an older HP Thin Client, such as a 5720. Using AstLinux on it's flash memory, 512K or 1 Gig, with 512K memory. The HP thin clients are available used, often quite inexpensive, and are already packaged. AstLinux can be remotely managed with the GUI,

Re: [asterisk-users] rasberry pi

On Wednesday 06 Jul 2016, John Novack wrote: > AstLinux can be remotely managed with the GUI, > which unlike other Asterisk GUI's the conf files are not modified by the > GUI and can be edited "by the book" AstLinux will NOT work with a Pi > though. It is not for the ARM processor. What stops it

[asterisk-users] Impossible to use any recent asterisk version with chan_sip

Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets. Nothing is written in the

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

Hi List, I solve this issue and I want share it with this community. The sng-tc-linux-1.3.8 package don't compile across Certified Asterisk. Only normal Asterisk like 11.22.0 version. We have this version in production with the D100 board. Working. Cheers -- GnuPG Key ID: 0x39BCA9D8

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

Barry Flanagan wrote: Joshua, This was actually reported in https://issues.asterisk.org/jira/browse/ASTERISK-25468 with backtraces. It appeared to have started in 13.5, as I tested all from 13.2 to 13.6 at the time Ah, that slipped my memory! Only the single person reporting it thus far.

Re: [asterisk-users] OpenSIPS or Kamailio based fronting for Asterisk?

Thanks! My big issue is that everything I find on a Google search is for chan_sip. I want chan_pjsip/realtime integration with Kamailio or OpenSIPS. It is not even clear if that has been done. On Wed, Jul 6, 2016 at 7:10 AM, Barry Flanagan wrote: > On 5 July 2016 at

[asterisk-users] Identify more demanding routine inside Asterisk

Hi List ! I'm facing a problem with the CPU consumption in Asterisk 11.22.0. I could decrease a lot of load, migrating both the astdb.sqlite3 and call recordings (with Monitor app) to a tmpfs mount in RAM (with noatime and nodiratime flags), manually spread each of the hardware interrupts

Re: [asterisk-users] how to read sip debug

Hi Thufir, The analysis of a SIP Debug depends on what the problem to be solved. If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

Leandro Dardini wrote: Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets.

Re: [asterisk-users] how to read sip debug

Another nice sip packet is sngrep Shows realtime the sip flows But i think you have to chk the asterisk answer in the dialplan logic to chk what context its hitting etc. בתאריך 6 ביולי 2016 10:05 PM,‏ "Steve Edwards" כתב: > On Wed, 6 Jul 2016, Victor Villarreal wrote:

Re: [asterisk-users] how to read sip debug

On Wed, 6 Jul 2016, Victor Villarreal wrote: If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or from where it is supposed to come

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

Leandro Dardini wrote: This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. And thank you for testing the release candidate so we can ensure the issue is fixed before

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Leandro 2016-07-06 21:46 GMT+02:00 Joshua Colp : > Leandro Dardini wrote: > >> Hello, >> I'd like to

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

Leandro Dardini wrote: This is a great news, thank you. I have open the issue, https://issues.asterisk.org/jira/browse/ASTERISK-26177 and added the relevant files, let me know if you need more info. Nothing else is needed from you! All the information has shown it's actually func_odbc. It's

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

On 6 July 2016 at 16:14, Joshua Colp wrote: > Leandro Dardini wrote: > >> Hello, >> I'd like to know if anyone of you is finding my same problems using any >> recent asterisk version, after 13.7 / 13.8 with chan_sip. >> >> If I use any recent asterisk version, after just few

Re: [asterisk-users] Impossible to use any recent asterisk version with chan_sip

Leandro Dardini wrote: Hello, I'd like to know if anyone of you is finding my same problems using any recent asterisk version, after 13.7 / 13.8 with chan_sip. If I use any recent asterisk version, after just few seconds asterisk completely locks up, stopping processing SIP/UDP packets.

Re: [asterisk-users] rasberry pi

A J Stiles wrote: On Wednesday 06 Jul 2016, John Novack wrote: AstLinux can be remotely managed with the GUI, which unlike other Asterisk GUI's the conf files are not modified by the GUI and can be edited "by the book" AstLinux will NOT work with a Pi though. It is not for the ARM processor.

Re: [asterisk-users] Function SHELL not registered

On Wednesday 06 Jul 2016, Michael Jepson wrote: > Adding live_dangerously did the trick. Thanks! But how dangerous is > Asterisk living now ? I must admit, still using an ancient Asterisk version, I didn't know about live_dangerously. But it sort of makes sense. It is somewhat dangerous to

Re: [asterisk-users] Function SHELL not registered

On Wed, Jul 6, 2016 at 4:05 AM, Michael Jepson wrote: > Adding live_dangerously did the trick. Thanks! But how dangerous is Asterisk > living now ? > > > >From README-SERIOUSLY.bestpractices.txt: === Avoid Privilege Escalations

[asterisk-users] rasberry pi

I'm debating between a cloud PBX or, perhaps, rasberry pi. For a SOHO, maybe three hardphones, rasberry pi would suffice? I would be amazed, but, if so, great. thanks, Thufir -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] what is a SIP invite, and who can issue them?

On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp wrote: > An INVITE is a request to set up a session, commonly referred to as a > call. Anything supporting SIP to establish calls uses INVITE to do so. > It's equivalent to picking up the phone and dialing a number. an INVITE would never be sent

Re: [asterisk-users] Function SHELL not registered

Adding live_dangerously did the trick. Thanks! But how dangerous is Asterisk living now ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston Sent: dinsdag 5 juli 2016 17:41 To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] what is a SIP invite, and who can issue them?

thufir wrote: On Wed, 29 Jun 2016 06:38:53 -0300, Joshua Colp wrote: An INVITE is a request to set up a session, commonly referred to as a call. Anything supporting SIP to establish calls uses INVITE to do so. It's equivalent to picking up the phone and dialing a number. an INVITE would

Re: [asterisk-users] rasberry pi

I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with Ubuntu Server 14.04. Works fine! :-) Frank On Wed, 2016-07-06 at 01:10 -0700, Thufir wrote: > I'm debating between a cloud PBX or, perhaps, rasberry pi. For a > SOHO, maybe three hardphones, rasberry pi would suffice? I

Re: [asterisk-users] rasberry pi

ok, that's really all I need to know. Of course, if anyone else wants to throw in their two cents, don't let me stop you :) -Thufir On Wed, Jul 6, 2016 at 1:36 AM, Frank Vanoni wrote: > I'm currently using Asterisk 11.7.0 on a Raspberry Pi 2 Model B with > Ubuntu