Re: [asterisk-users] Which CDR processing for high load ?

2018-02-22 17:12 GMT+01:00 Dovid Bender : > Have you looked at the proc limits for the Asterisk PID? > > > No I haven't. What shall I look for, exactly ? The only thing I configured for performance is asterisk.conf 's maxfiles parameter. --

[asterisk-users] Which CDR processing for high load ?

Hello, I'm load testing a new Asterisk 13 system (Debian Stretch, packaged asterisk). One system writes CDR though an ODBC connection to a local Postgres database over the LAN. When sending 50 new calls per second with SIPp, I'm seeing one system outputs : taskprocessor.c: The

Re: [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13

:00 Olivier <oza.4...@gmail.com>: > Hi, > > Reading this old thread, may I ask if keeping hangup handlers from > updating CDR values still enforced in Asterisk 15 ? > If positive, would it be very complex to add in Asterisk, a configuration > option allowing a syst

Re: [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13

Hi, Reading this old thread, may I ask if keeping hangup handlers from updating CDR values still enforced in Asterisk 15 ? If positive, would it be very complex to add in Asterisk, a configuration option allowing a system administrator to list in cdr.conf, the CDR fields allowed to be updated in

[asterisk-users] [OT] Gigaset N510IP provisionning

Hello, Has someone met success in Gigaset N510IP DECT base station provisionning ? If positive, could you describe a bit which files you had to create on (HTTP) provsionning server ? Best regards -- _ -- Bandwidth and

Re: [asterisk-users] [OT] How to use audio files with SIPp

Thank you very much George for replying. 2018-02-09 14:39 GMT+01:00 George Joseph <gjos...@digium.com>: > > > On Fri, Feb 9, 2018 at 6:27 AM, Olivier <oza.4...@gmail.com> wrote: > >> Hello, >> >> SIPp's PCAP play feature can replay pre-recorded a

[asterisk-users] How to add SNMP support to packaged asterisk on Debian stretch

Hello, If I'm not mistaken SNMP support is missing in Debian Stretch packaged Asterisk while this support is present in either Jessie or Buster (looking at [1] or equivalent pages). Is it something that can be worked around or shall I fear a major obstacle when re-packaging my own asterisk

[asterisk-users] [OT] How to use audio files with SIPp

Hello, SIPp's PCAP play feature can replay pre-recorded audio stream towards destination (see [1]). Doc mentions tcpdump and Wireshark as tools to record such RTP streams without further details. Looking at SIPp 3.2 source archive, I found PCAP samples in a pcap/ directory. Sample

Re: [asterisk-users] pjsip trunking configuration issue

@Kevin: Were such invalid endpoint parameters settings reported on Asterisk CLI ? My system reported: Could not find option suitable for category 'asterisk8' named 'foo' at line 15 of /etc/asterisk/pjsip.conf when I added ( at line 15 of /etc/asterisk/pjsip.conf): foo=bar 2018-02-08 14:48

[asterisk-users] Features.conf and variable length DTMF sequences

Hello, I have an Asterisk 13-enabled system. 1. Using features.conf application map (or something else), is it possible to define a single map matching several DTMF sequences, such as in the imaginary example bellow ? features.conf: foobar => _*123.,peer,Gosub,"foobar,s,1" _*123. would match

[asterisk-users] Is Asterisk 11's chan_sip able to send RTCP reports ?

Hello, On a lab setup, I can see an Asterisk 11 system is correctly receiving and displaying (sip show channelstats) incoming RTCP reports but not any report to the other end. Searching through *.sample files does show much. This highlighted I still have a lot to learn on RTCP. My setup is:

Re: [asterisk-users] Can't install package asterisk-dbgsym on Stretch [SOLVED]

Hi, 2017-12-14 16:28 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Fri, Dec 08, 2017 at 06:11:47PM +0100, Olivier wrote: > > Hello, > > > > On a fresh Debian Stretch setup, I have: > > $ cat /etc/apt/sources.list.d//dbgsym.list > > deb http:

Re: [asterisk-users] DTMF emulation with SIP INFO and direct media

Hello Jean, 1. Can you describe a bit further how both ends of the above call were both made of and configured ? DTMF receiving is Asterisk/SIP channel but which version ? Is the other end a SIP phone or a SIP trunk ? 2. Do you observe such behaviour in a one-to-one setup (one end emits, the

Re: [asterisk-users] [OT] Overview of Homer installation on Debian Stretch

ity and Communication by Commend *FN 178618z | LG Salzburg > > > > *Von:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *Im Auftrag von *Olivier > *Gesendet:* Dienstag, 12. Dezember 2017 16:59 > *An:* Asterisk Users Mailing List - Non-Com

Re: [asterisk-users] General Kernel practices on CentOS

t; Ron > > On 14/12/2017 10:38 AM, Olivier wrote: > > Hello, > > I'm used to install Asterisk on Debian stable platforms. > > A customer is asking how I would proceed on a CentOS platform. > > After a short research (see [1] as an example), I'm wondering what are &

[asterisk-users] General Kernel practices on CentOS

Hello, I'm used to install Asterisk on Debian stable platforms. A customer is asking how I would proceed on a CentOS platform. After a short research (see [1] as an example), I'm wondering what are general kernel practices on CentOS regarding Asterisk and when targeting stability: - Is it

[asterisk-users] Explain HangupCauseClear() and HANGUPCAUSE_KEYS behaviour

Hello, I'm giving HangupCauseClear() a try on a Debian Stretch / Asterisk 13.18.3 stack. My dialplan is: exten = 1234,1,Set(CHANNEL(hangup-handler-push)=myhandler,s,1) same = n,Dial(SIP/foo/1234) same = n,Gosub(myhandler,s,1) same = n,HangupCauseClear() same = n,Dial(SIP/bar/1234) [myhandler]

Re: [asterisk-users] Showing CallerID on multiple phones

When a phone supervises an other phone (with BLF and NOTIFY/SUBSCRIBE) it gets from Asterisk everything it needs to do what you're after. Some phone vendors should support config option to add screen display along BLF blinking Some might even enhance this with a short audio notification.

[asterisk-users] [OT] Overview of Homer installation on Debian Stretch

Hello, I've discovered homer-api-postgresql and homer-api-mysql packages in Stretch repo. I'm not sure I understand how Homer-API relates to Homer. My questions are: 1. What is the simplest available installation option to install Homer on a dedicated box, this dedicated box gathering data

[asterisk-users] How to generate TLS certificates on Debian

Hello, On a Debian Stretch box with packaged asterisk (asterisk 13.14.1), which tool can I use to generate TLS certificates ? Doc on [1] mentions an ast_tls_cert script (from contrib/script) which is not installed by Debian package. Is there some equivalent tools from general purpose packages

[asterisk-users] Can't install package asterisk-dbgsym on Stretch

Hello, On a fresh Debian Stretch setup, I have: $ cat /etc/apt/sources.list.d//dbgsym.list deb http://debug.mirrors.debian.org/debian-debug/ stretch-debug main # apt-get update ... # apt-get install asterisk gdb # apt-get -s install asterisk-dbgsym ... asterisk-dbgsym : Depends: asterisk (=

Re: [asterisk-users] How to read or write Geolocation (RFC6442) data in SIP/PJSIP messages ?

Thank you both for replying ! 2017-12-08 16:02 GMT+01:00 Joshua Colp : > On Fri, Dec 8, 2017, at 10:58 AM, Jean Aunis wrote: > > Hello, > > > > As far as I know there is no way to read or write the INVITE's body, > > neither with chan_sip nor chan_pjsip. > > This is correct.

[asterisk-users] Explain how to maintain a compiled from source Asterisk instance ?

Hello, When compiling Asterisk from source, the classical ./configure, make and make install commands are issued. If a vulnerabilty is found within Asterisk code, then Asterisk source code is patched and depending on what files were touched parts or all of above commands need to be re-issued.

Re: [asterisk-users] How can I check backtrace files ? [SOLVED]

2017-12-07 15:50 GMT+01:00 George Joseph <gjos...@digium.com>: > > > On Wed, Dec 6, 2017 at 11:13 AM, Olivier <oza.4...@gmail.com> wrote: > >> >> >> 2017-12-06 15:52 GMT+01:00 George Joseph <gjos...@digium.com>: >> >>> >>>

[asterisk-users] How to read or write Geolocation (RFC6442) data in SIP/PJSIP messages ?

Hello, I'm having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length): INVITE sips:b...@biloxi.example.com SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74bf9 Max-Forwards: 70 To: Bob

Re: [asterisk-users] How can I check backtrace files ?

2017-12-06 15:52 GMT+01:00 George Joseph <gjos...@digium.com>: > > > On Tue, Dec 5, 2017 at 9:20 AM, Olivier <oza.4...@gmail.com> wrote: > >> Hello, >> >> I carefully read [1] which details how backtrace files can be produced. >> >> Maybe

[asterisk-users] How can I check backtrace files ?

Hello, I carefully read [1] which details how backtrace files can be produced. Maybe this seems natural to some, but how can I go one step futher, and check that produced XXX-thread1.txt, XXX-brief.txt, ... files are OK ? In other words, where can I find an example on how to use one of those

Re: [asterisk-users] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?

+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkinis...@gmail.com>: > Hello Olivier, > > I may be incorrect but I don't believe you can hint on a mailbox like > that. > > I've always used custom device states and dialplan logic for my

[asterisk-users] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?

Hello, I'm trying to supervise an existing Voicemail box with a BLF button on Debian's asterisk 13.14.1 system. I mostly found this [1] document. I added in a context a line like: exten = *7000,hint,MWI:31@default With "core show hints", I can read this: *7000@subs : MWI:31@default

[asterisk-users] Is it safe to configure SIP/Registry entries on a passive asterisk node ?

Hello, I've seen that Asterisk stores in ASTDB entries like: /SIP/Registry/spa3102 : 192.168.64.207:5060: 3600:7013:sip:spa3102@192.168.64.207:5060 1. My understanding is that any peer that sent to Asterisk a REGISTER message has such entry set. So having these

Re: [asterisk-users] How to log missing RTP packets ?

g logging is quite verbose on console. Having an option to have a more concise logging would be welcome. 2017-11-10 15:38 GMT+01:00 Joshua Colp <jc...@digium.com>: > On Fri, Nov 10, 2017, at 10:34 AM, Olivier wrote: > > Hello, > > > > When a call is starting, Aster

[asterisk-users] How to log missing RTP packets ?

Hello, When a call is starting, Asterisk starts sending and receiving RTP packets. Each packet has a sequence number. 1. Instead of logging everything as rtp set debug is currently doing, is there a way to only log: - the sequence number of the first received packet, - any missing or misplaced

[asterisk-users] How to use console channels ?

Hello, To troubleshoot Voice Quality issues and automate testing, I'm planning to use Asterisk as a softphone alternative (with better scripting capabilities) on a Linux 4.10-enabled Ubuntu laptop. This laptop has two audio jacks with a compatible headset. I'm testing console channel with lines

Re: [asterisk-users] Remote Phonebook with Thomson ST2022

Hi, Have you read about ST2030 remote phonebook examples ? As both phones share the same engineering and ST2030 was quite successfull, years ago, that may be worth looking at ST2030 documentation. Best regards 2017-10-25 19:45 GMT+02:00 Luca Bertoncello : > Hi list! > >

Re: [asterisk-users] Measuring total end-to-end latency

Hi, I don't have a direct answer, but I've read several times about purposely customized system over the PSTN, echoing incoming incoming audio to produce metrics when troubleshooting call quality. I alse remember a thing called Recqual targeting the same goal. I hope this helps 2017-10-31

[asterisk-users] Many tests from TestSuite fail with "Asterisk 127.0.0.1 received error: FRACK!"

Hello, Thanks to Tzafrir and George help, I could run Asterisk TestSuite for the first time on a fresh Debian Stretch setup. TestSuite is installed with "apt-get install asterisk-testsuite" and Asterisk itself is stopped (with "service asterisk stop") before executing (as root)

Re: [asterisk-users] How tu run runtests.py on Debian Stretch ? [SOLVED]

2017-10-31 12:14 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Tue, Oct 31, 2017 at 12:07:57PM +0100, Olivier wrote: > > Hello, > > > > I'm giving asterisk-testsuite package a try on a fresh Debian Stretch > setup. > > > > I've got this: >

[asterisk-users] How tu run runtests.py on Debian Stretch ?

Hello, I'm giving asterisk-testsuite package a try on a fresh Debian Stretch setup. I've got this: # /usr/share/asterisk-testsuite/runtests.py Traceback (most recent call last): File "/usr/share/asterisk-testsuite/runtests.py", line 24, in from asterisk.version import AsteriskVersion

Re: [asterisk-users] ASR Suggestions for small dictionnary (<1000 entries) lookup in France/french

t they must say one digit at a time, in > which case, easy! Pick any free speech recognition you want. > 2: Try and handle "natural language" ways of speaking numbers, in > which case be prepared for a lot of debugging and learning! > > I could be wrong - if anyone knows, ple

[asterisk-users] ASR Suggestions for small dictionnary (<1000 entries) lookup in France/french

Hello, I'm in the early stages of designing an Emergency calling service IVR application. The IVR application asks simple one or two questions like "which is the postal code of the area you are currently calling from ?" "Is the correct ?". The expected values are a 5-digits number like

[asterisk-users] Load testing with media in batch mode

Hello, I am currently tasked on how to load test both signal and media from a couple of Asterisk machines which are doing corporate SIP trunking (no phone endpoint). If that matters, ecah machine will host debian Stretch, Asterisk 13 with either classic SIP or PJSIP. For instance, I can

[asterisk-users] Asterisk and Virtual routing and forwarding (VRF)

Hello, Reading [1], Asterisk's PJSIP requires a unique IP+port combination when configuring multiple transport instances. Linux now supports Virtual routing and forwarding (VRF) (see [2]). Is there a way to set a given PJSIP transport to use a given interface and doing so, brings VRF-awareness

Re: [asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

; Yes certainly, that's an interesting solution that also brings other benefits, its main limitation being my current of OpenSiPS experience. Thanks for replying > > > > On Wed, Sep 6, 2017 at 5:39 AM, Olivier <oza.4...@gmail.com> wrote: > >> Hello, >> >> I'm

[asterisk-users] Asterisk 13.X with multiple IP addresses: Can I force a given chan_sip peer to a given IP address ?

Hello, I'm quite sure this question has already be asked previously but before diving into it with a lab setup, I would like to re-ask here the thereafter question. I've got a bunch of very old Asterisk boxes (lastest Asterisk version is 1.6.1.X), all belonging to the same network, I would like

[asterisk-users] Improvement of PJSIP dtmf_mode description

Hello, While debugging a SIP trunk with an Avaya IPO, I noticed that wiki's PJSIP dtmf_mode at [1] includes: "This setting allows to choose the DTMF mode for endpoint communication. rfc4733 - DTMF is sent out of band of the main audio stream. This supercedes the older RFC-2833 used within

[asterisk-users] Planet VIP156PE sample config file

Hello, I've just discovered this Planet VIP156PE ATA (see (1]). It has the unique feature of being PoE powered. Its web administration application mentions provisionning capabilty but I could not find any documentation about it. Do you have any (anonymized) sample config file ? I'm planning to

Re: [asterisk-users] CentOS7: How to debug SEGV when asterisk starts with autoload=yes ?

back from a business trip. 2017-06-12 15:29 GMT+02:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Mon, Jun 12, 2017 at 10:36:21AM +0200, Olivier wrote: > > Hello, > > > > I was tasked to install Asterisk 13.16.0. from source on a CentOS7 > platform. > > >

[asterisk-users] CentOS7: How to debug SEGV when asterisk starts with autoload=yes ?

Hello, I was tasked to install Asterisk 13.16.0. from source on a CentOS7 platform. For that purpose, I used an unmaintened script of mine, written 10 monthes ago, and I was surprised to get segmentation violations whenever I ran "asterisk -cvvv -U asterisk". Usually, my

[asterisk-users] OT: Explain where mailing list bouncing comes from ?

Hello, I'm a faithful reader of this mailing list, for several years now. Lately, I'm receiving emails asking me to re-enable my list subscription due to "excessive bouncing". What does this exactly mean and why am I receiving this ? Beside re-enabling my subscription, what can I do to improve

Re: [asterisk-users] Asterisk 13 attended transfer alcatel

How are both machines connected to each other ? Through a SIP trunk ? A TDM one ? 2017-06-09 9:59 GMT+02:00 Jason TOMLINSON : > Hello, > > > > Since upgrading from asterisk 11 to asterisk 13 (I have tested up to the > latest 13.16.0 release), we have a problem with

Re: [asterisk-users] Working around missing libmyodbc in Debian Stretch

ursday 08 Jun 2017, Olivier wrote: >> >>> Hello, >>> >>> I'm building a new Asterisk system from source on Debian Stretch. >>> My building script fails as package libmyodbc is currently missing from >>> Debian Stretch repo. >

[asterisk-users] Working around missing libmyodbc in Debian Stretch

Hello, I'm building a new Asterisk system from source on Debian Stretch. My building script fails as package libmyodbc is currently missing from Debian Stretch repo. Is there a work around this without leaving MySQL/MariaDB galaxy ? Best regards --

Re: [asterisk-users] Backport of Stretch's asterisk.service file into Jessie: successful start not detected by systemd [SOLVED]

Basically, adding libsystemd-dev on Jessie before recompiling (./configure, make, ...) allowed Asterisk to notify systemd it has successufully started. For reference, please note this feature requires Asterisk 13.12.0 and above. Thank you very much, Tzafrir, for your help ! --

[asterisk-users] Backport of Stretch's asterisk.service file into Jessie: successful start not detected by systemd

Hello, I've been tasked to enable automatic Asterisk restart on failure on a Jessie platform (running latest Asterisk 13.15.0). I build a dedicated Jessie VM on which I installed Asterisk from source. I configured a couple of files in /etc/asterisk directory. I positively checedk that with

[asterisk-users] Asterisk 13.14.0. Debugging DTMF issues

Hello, I'm working on a (PJ)SIP trunking Asterisk machine with which I'm facing issues with DTMF. Installed version is 13.14.0. 1. In outbound calls SDP, I'm seeing these kind of lines: a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 I would expect events to range from 0 to 15, not to 16, as

Re: [asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

2017-03-29 17:28 GMT+02:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Wed, Mar 29, 2017 at 05:18:18PM +0200, Olivier wrote: > > Hello, > > > > After reading [1] (in french), I would be very happy if I could get > answers > > to: > > > > 1. D

Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]

Basically, in some countries, regulation says: "caller MUST know in advance what call would given the number (s)he dialed". If regulation also states that waiting time shall not charged or shall not charged more than a local/national/whatever call, then an alternative is to separate

[asterisk-users] Questions regarding asterisk-opus package in Debian Stretch repo and Opus in general

Hello, After reading [1] (in french), I would be very happy if I could get answers to: 1. Does this 13.7+20161113-3 package version has any relation with asterisk's version it complements ? Current asterisk version in repo is 13.14.0. Does this 13.7 complies with it ? 2. From package

Re: [asterisk-users] How to have callers not being billed when in waiting queue ? [SOLVED]

any > until you send them the amount of time that should be billed for a specific > call. > > > Your best choice will be, that - if you ever get those regulations - you > should rely on what your telephone number provider tells you to do ;-) > > > Greetings > Max > &g

[asterisk-users] How to have callers not being billed when in waiting queue ?

Hello, In France, years ago, there was some discussions about a new regulation forcing some providers to not charge anything to callers while those are waiting for a call center agent to become available. Once caller and agent are on call with each other, nominal charging applies. No matter if

[asterisk-users] Questions regarding Dial's D option

Hello, I'm currently playing with Application Dial D option. This option is documented with: D([called][:calling[:progress]]): Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The DTMF string is sent to the called party, and

Re: [asterisk-users] Asterisk 13: is CALLERID(num-pres) readable ? [SOLVED]

Hello, 2017-03-17 15:05 GMT+01:00 Olivier <oza.4...@gmail.com>: > Hello, > > From a 13.14.0 system: > > same = n,Verbose(0,1-CALLERID(num-pres) is ${CALLERID(num-pres)}) > same = n,Set(CALLERID(num-pres)=prohib) > same = n,Verbose(0,2-CALLERID(num-pres) is now ${CALLER

Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?

ere can Dahdi 2.11.1 Changelog file be found ? In http://downloads.asterisk.org/pub/telephony/ ? Best regards 2017-03-20 16:25 GMT+01:00 Olivier <oza.4...@gmail.com>: > > > 2017-03-14 15:26 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > >> On Tue, Mar 14, 2017 at

Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?

2017-03-14 15:26 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Tue, Mar 14, 2017 at 02:58:07PM +0100, Olivier wrote: > > 2017-03-14 13:08 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > > > > > On Tue, Mar 14, 2017 at 11:10:57AM

[asterisk-users] Asterisk 13: is CALLERID(num-pres) readable ?

Hello, >From a 13.14.0 system: same = n,Verbose(0,1-CALLERID(num-pres) is ${CALLERID(num-pres)}) same = n,Set(CALLERID(num-pres)=prohib) same = n,Verbose(0,2-CALLERID(num-pres) is now ${CALLERID(num-pres)}) I would expect to read "2-CALLERID(num-pres) is now prohib" but I get

Re: [asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?

2017-03-14 13:08 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Tue, Mar 14, 2017 at 11:10:57AM +0100, Olivier wrote: > > Hello, > > > > After all these years installing from source, I'm giving Dahdi package > > installation a try on a recent Stretch bo

[asterisk-users] How to install and configure Dahdi from Debian Stretch repo ?

Hello, After all these years installing from source, I'm giving Dahdi package installation a try on a recent Stretch box. Google over the web, I didn't find too many doc on this topic. 1. Is this one [1] up-to-date ? Reading Stretch I would say a single asterisk-dahdi would be enough to install

Re: [asterisk-users] Which tool to automatically restart Asterisk ?

2017-02-21 14:09 GMT+01:00 Tahir Almas : > Why not to use Fail2ban https://www.voip-info.org/ > wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk > > How would fail2ban detect that Asterisk needs to be restarted ? > > > *Tahir Almas* > > Managing Partner > ICT

[asterisk-users] Which tool to automatically restart Asterisk ?

Hello, Years ago, I used Monit to monitor Asterisk and restart it whenever it failed. Now, I wonder which tool I should pick for an Debian 8 (current) or CentOS 7 (future) environment. The main reason I'm looking for this tool is to avoid as much as possible, current 5 minutes delay between

Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

2017-02-17 14:39 GMT+01:00 George Joseph : > > > > If asterisk was compiled with DEBUG_THREADS, > Would you then advise to run an Asterisk server in production with DEBUG_THREADS enabled ? Page [1] does not mention to do so nor to avoid it. In the production environment I'm

Re: [asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

gt; > On Tue, Feb 14, 2017 at 2:51 PM, George Joseph <gjos...@digium.com> wrote: > >> >> >> On Tue, Feb 14, 2017 at 10:21 AM, Olivier <oza.4...@gmail.com> wrote: >> >>> Hello, >>> >>> I've got a 13.13.1 system using PJSIP stack on d

Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

2017-02-16 19:33 GMT+01:00 Matthew Jordan <mjor...@digium.com>: > > > On Thu, Feb 16, 2017 at 9:05 AM, Olivier <oza.4...@gmail.com> wrote: > >> >> >> 2017-02-16 14:27 GMT+01:00 Joshua Colp <jc...@digium.com>: >> >>> On Thu, Feb 16,

Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

2017-02-16 14:27 GMT+01:00 Joshua Colp <jc...@digium.com>: > On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote: > > Hello, > > > > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP > > hardphone. > > > > When a phone has enabled thi

[asterisk-users] How to read or relay SIP PUBLISH messages ?

Hello, I'm currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone. When a phone has enabled this feature, it would send a SIP PUBLISH to its SIP Server letting this server dispatch to whatever is needs to. These messages are sent during calls but may also be sent when a call is

Re: [asterisk-users] [OT] Downloading Recqual [SOLVED]

2017-02-16 12:21 GMT+01:00 A J Stiles <asterisk_l...@earthshod.co.uk>: > On Thursday 16 Feb 2017, Olivier wrote: > > Hello, > > > > While googling, I've just discovered Recqual. > > If I'm not mistaken, project's sourceforge site [2] does not host any > > s

[asterisk-users] [OT] Downloading Recqual

Hello, While googling, I've just discovered Recqual. If I'm not mistaken, project's sourceforge site [2] does not host any source or binary. Is there an alternative location to download this ? Suggestions ? Best regards [1] http://blog.krisk.org/2008/12/introducing-recqual.html [2]

[asterisk-users] [OT] How to collect VQ RTCP-XR ? What can you hope from VQ RTCP-XR reports ?

Hello, I've read a couple of pages about VQ RTCP-XR (RFC 6035) which is implemented in some SIP hardphones. Do you have any experience to share about using VQ RTCP-XR ? Does it help to hightlight issues when those issues come from a provider of your ITSP ? Which collector do you use to collect

[asterisk-users] Advices when Asterisk segfaults and nothing useful in logs

Hello, I've got a 13.13.1 system using PJSIP stack on debian Jessie. It runs from 50 to 100 simultaneous calls (so 100 to 200 PJSIP channels) all day long. >From time to time, roughly meaning once a month, it segfaults with lines (from dmesg -T output) like this: asterisk[1160]: segfault at

Re: [asterisk-users] Asterisk 13.13.1

arifi > *Sent:* Saturday, January 28, 2017 5:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Asterisk 13.13.1 > > > > > > On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4...@gmail.com> wrote: > > What did you exactly upgade ?

Re: [asterisk-users] Asterisk 13.13.1

What did you exactly upgade ? Asterisk only ? Asterisk and OS ? How did you installed Asterisk 1.8 and 13 ? From source or from package ? I would be curious to see what would happen after downgrading back to 1.8. 2017-01-24 21:03 GMT+01:00 Motty Cruz : > Hello, I recently

Re: [asterisk-users] Understanding how LLDP works with DHCP [SOLVED]

2017-01-19 18:19 GMT+01:00 Jose Flores Galicia <floj...@gmail.com>: > 2017-01-19 4:09 GMT-06:00 Olivier <oza.4...@gmail.com>: > >> Hello, >> >> For years, I used to configure SIP phone VLAN membership through a DHCP >> server. >> >> Here a

[asterisk-users] Understanding how LLDP works with DHCP

Hello, For years, I used to configure SIP phone VLAN membership through a DHCP server. Here are the details: - I dedicate a LAN port on a switch to voice VLAN - somewhere else, I configure a DHCP server to serve LAN addresses within voice VLAN - any other switch port connected to an other DHCP

Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

2017-01-18 19:35 GMT+01:00 Israel Gottlieb : > snom could get lots of configuration options thru sip notify > I didn't know that. > i once tried updateing the display name on hot desking but ran in to his > problem of having to add it to sip conf staticly > Here, you mean

Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

ayed message is not very attractive. > Thanks; > > John V. > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *Olivier > *Sent:* Wednesday, January 18, 2017 08:54 AM > *To:* Asterisk Users Mailing L

Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

n a side note, to my knowledge, SIP NOTIFY is currently used: - for MWI, - for rebooting/triggering SIP phones - for reconfiguring SIP phones (I've just discovered this usage). Do you know other usages ? > > > -Thufir > > On Mon, 16 Jan 2017, Olivier wrote: > > Thinking ov

Re: [asterisk-users] How to send SIP_NOTIFY messages with variable content ?

Thinking over my previous, I wonder if sipsak could be used to send outgoing SIP NOTIFY messages. Would both Asterisk and sipsak be able to share networks resources ? Thoughts ? 2017-01-16 14:10 GMT+01:00 Olivier <oza.4...@gmail.com>: > Hello, > > One common mean to remotely co

[asterisk-users] How to send SIP_NOTIFY messages with variable content ?

Hello, One common mean to remotely configure a phone is to send it some XML data using HTTP. Of course, this XML data is vendor specific but thanks to Asterisk multiple tools, it is quite easy to customize your dialplan to both build and send this specific XML data. I have just discovered one

Re: [asterisk-users] Does HEP require PJSIP or does it also works with SIP ?

Hello, I filed issue https://issues.asterisk.org/jira/browse/ASTERISK-26717 for this. Thanks for helping 2017-01-03 16:17 GMT+01:00 Joshua Colp <jc...@digium.com>: > On Tue, Jan 3, 2017, at 11:04 AM, Olivier wrote: > > Hello, > > > > On a newly built Asterisk 13.13

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

and rebuilding everything from scratch won't help me to learn anything). Thank you all very much for your previous help. 2017-01-10 14:24 GMT+01:00 George Joseph <gjos...@digium.com>: > > > On Mon, Jan 9, 2017 at 5:15 PM, Olivier <oza.4...@gmail.com> wrote: > >> Hello, &g

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

e (such as --prefix). 2017-01-10 10:31 GMT+01:00 A J Stiles <asterisk_l...@earthshod.co.uk>: > On Tuesday 10 Jan 2017, Olivier wrote: > > Hello, > > > > I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. > > > > I followed this: > > cd /usr

Re: [asterisk-users] Can't comile bundled PJSIP on CentOS 7

2017-01-10 5:30 GMT+01:00 Anton Teyhrib <teih...@gmail.com>: > Maybe libsrtp package is missing on your machine? > libsrtp-devel was installed and I could check it was installed with menuselect (Resources Modules/res_srtp) > 10 янв. 2017 г. 5:16 AM пользователь &

[asterisk-users] Can't comile bundled PJSIP on CentOS 7

Hello, I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. I followed this: cd /usr/src wget ... asterisk-13.13.1.tar.gz tar zxf asterisk-13.13.1.tar.gz cd asterisk-13.13.1 ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr" ./configure ${ASTERISK_CONFIGURE}

[asterisk-users] Does HEP require PJSIP or does it also works with SIP ?

Hello, On a newly built Asterisk 13.13.1 system, I can't make HEP work with chan_sip (though I could make it work with PJSIP on another instance). Looking either at [1] or hep.conf, I can't see anything meaning HEP requires PJSIP. Before diging deeper, may I simply ask if HEP requires PJSIP or

Re: [asterisk-users] Asterisk hep.conf

Hello, Reading this thread, may I ask if you could get this to work ? Regards 2016-06-29 6:29 GMT+02:00 Annus Fictus : > hello, > > I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server. > > My hep.conf Asterisk configuration is: > > [general] > enabled = yes >

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

Do you have any LLDP or CDP enabled anywhere ? 2016-12-21 19:50 GMT+01:00 Victor Villarreal : > Hi Yves, > > Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC > of the phone. Maybe with the snom this not happen because your switch don't > see the MAC

[asterisk-users] [OT] Outbound on SPA3102 FXO stopped to work. Where to look at ?

Hello, I'm currently facing a quite strange issue. On a customer location, an old SPA3102 suddenly stopped to work a couple of days ago. More precisely, calls still come in but I can't dial out to PSTN. This box worked for several years for oubound and inbound calling. My setup is: Asterisk 11

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

2016-12-19 16:26 GMT+01:00 Yves : > Hi, > > I am pulling my hair for days now... > > I can´t get a PolyCom SoundStation IP 6000 (Conferencephone) to register > with my Asterisk. > > There are no SIP Packets arriving at my asterisk at all... and it has > nothing to do with a

Re: [asterisk-users] Asterisk installation script on CentOS7 with systemd [SOLVED]

2016-12-19 16:11 GMT+01:00 Jean Aunis <jean.au...@prescom.fr>: > Le 19/12/2016 à 15:54, Olivier a écrit : > > Hello, > > For a new project, I'm adapting existing installation script to CentOS 7. > I must admit I don't understand how to adapt things to systemd. > > H

[asterisk-users] Asterisk installation script on CentOS7 with systemd

Hello, For a new project, I'm adapting existing installation script to CentOS 7. I must admit I don't understand how to adapt things to systemd. Here are my questions: 1. I don't see any systemd sub-directory in asterisk-13.13.1/contrib. Do you think such directory and matching Makefile target

Re: [asterisk-users] What to do when changing from one asterisk version to another ?

2016-12-13 16:19 GMT+01:00 Tzafrir Cohen <tzafrir.co...@xorcom.com>: > On Thu, Dec 08, 2016 at 06:23:15PM +0100, Olivier wrote: > > Hello, > > > > I'm compiling Asterisk from source on Debian systems. > > > > I'm currently writing a script I'm plannin

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