search for: limbani

Displaying 20 results from an estimated 56 matches for "limbani".

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2012 Jun 03
2
Caller ID : FSK ETSI or FSK US
Hello, All :) Regarding to incoming caller ID on PSTN line, which one is best supported by asterisk: is it FSK ETSI or FSK US? I bought some caller ID converter hardware (convert DTMF to FSK and vice versa) but still asterisk can not detect it. The converter has a switch FSK ETSI or FSK US This is what I put in /etc/asterisk/chan_dahdi.conf ... cidsignalling=bell cidstart=ring ... If after
2016 Feb 17
2
1000 analogue lines with asterisk
On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > Sangoma 50 port FXS Thanks. Will I now stack 20 boxes in order to achieve the 1000 FXS lines? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160217...
2016 Feb 17
2
1000 analogue lines with asterisk
Thanks Mitul, The server spec is okay but I need information on the fxs hardware to use. Regards On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani <mitul at enterux.in> wrote: > Quad core Xeon with 4GB ram > On Feb 17, 2016 12:32 PM, "Goke Aruna" <goksie at gmail.com> wrote: > >> Hello all, >> Can someone recommend what hardware to use for a 1000 analogue line >> capacity asterisk PABX? >&g...
2013 Mar 31
0
asterisk-users Digest, Vol 104, Issue 53
...please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. ISDN- E1 PRI module in network side signaling (Dimitar Dimitrov) > 2. Re: ISDN- E1 PRI module in network side signaling (Mitul Limbani) > 3. Re: ISDN- E1 PRI module in network side signaling > (Tony Mountifield) > 4. Re: ISDN- E1 PRI module in network side signaling > (Dimitar Dimitrov) > > > ---------------------------------------------------------------------- > > Message: 1 > Date...
2016 Apr 08
2
Recommendations for free virtual server tech and Asterisk?
If you want to use dahdi dummy driver inside asterisk for timer then this is possible with openvz based container virtualization. We have tested vicidial in this mode for 5-10 agents and it worked well. Mitul Limbani On Apr 8, 2016 8:52 AM, "Pete Mundy" <pete at fiberphone.co.nz> wrote: > List, > > Might as well throw my hat in the ring! > > I can't say it's the 'best' way to do it, but I've been running Asterisk > VMs inside the free 'VirtualBox' s...
2012 May 07
6
using Wifi smartphones as SIP clients
All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.
2016 Feb 17
2
1000 analogue lines with asterisk
...ght port T1 cards, or with eleven/twelve > quad T1 cards. I would distribute across two, three, or even four servers > for redundancy/resiliency and load balancing. > > -Harry > > > On 02/17/2016 12:16 AM, Goke Aruna wrote: > > > On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote: > >> Sangoma 50 port FXS > > > > Thanks. > Will I now stack 20 boxes in order to achieve the 1000 FXS lines? > Regards > > > > > -- > _____________________________________________________________________ > -- Bandwid...
2015 Feb 22
1
dialplan contexts syntax and terminology
On Sun, 22 Feb 2015 08:32:26 +0530, Mitul Limbani wrote: > READ READ READ .... I know, I have the 4th edition and I've been reading it. Personally, I find it more general than specific, but I'll go back through that chapter, absolutely. thanks, Thufir
2013 Mar 15
2
app_rtsp.c ported to Asterisk 11.x
Hi, If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC video from one machine to another machine running Linphone. Contact me at this e-mail address robkrakora at messagenetsystems.com for source code. Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032
2020 May 25
2
Asterisk : CDR Analyzer Updated
Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7 system, since it won't work with the newer PHP7. A friend of mine is learning PHP7
2012 Jun 02
1
Asterisk pickup call on first ring
Hello, Currently my asterisk system pickup incoming call after 3 or 4 rings. How can I ask it to answer the call on the first ring? I put immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no different. Thanks in advance :) BR, Anam -- Sent from my mobile device
2015 Oct 05
2
Fwd: Sublime Text License Key
The company making sublime text gets few thousands of dollars of notional loss :) On Oct 5, 2015 8:45 PM, "Steve Howes" <steve-lists at geekinter.net> wrote: > Wonder what happens when an entire mailing list tries to use that key?... > > On 05/10/15 15:28, Optical Phoenix wrote: > > ---------- Forwarded message ---------- > From: *Sublime HQ Pty Ltd* <sales at
2015 Apr 07
2
OpenVZ with asterisk 13
On 04/07/2015 10:48 AM, Johan Wilfer wrote: > Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev: >> Dear all, >> >> Is anyone has experience making Asterisk server with virtual server >> OPEN-VZ (in proxmox 3.4 box) ? >> >> My boss want to build a production server with it, and it will have +/- >> 300 sip user (concurrent call maybe < 150 call) >>
2015 Feb 22
2
dialplan contexts syntax and terminology
I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk
2015 Jun 22
2
Product CDR/Queue/Meetme
Hello, ? I am interested, too. ? Att, Welinghton Citando Mitul Limbani <mitul at enterux.in>: > Hey Helvio, > > Would like to check it out as well. > > Do email me, > > Mitul > On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > >> Gentleman, >> >> Moderators, i don...
2012 Jul 24
5
DAHDI problems
Is a normal functionality? when I do #dahdi_cfg -vvvvvv In my Asterisk console shows this.... [Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 If I do this a lot of times...then [Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [Jul 24
2015 Apr 07
1
OpenVZ with asterisk 13
I have several large customers (200+ extensions) running on vSphere without issue. Not sure about OpenVZ, thought. 2015-04-07 11:36 GMT-03:00 Mitul Limbani <mitul at enterux.in>: > Show him this freaking thread, or else ask him to prove it otherwise. > > We all here have decades of exp dealing with asterisk. > > Mitul > On 07-Apr-2015 7:27 PM, "Ikka Tirtawidjaja" <ikka.tirta at gmail.com> wrote: > >> D...
2013 Feb 24
2
AEL Macro are evil :-)
I just discover an "hidden" problem with AEL macro I want to have your feedback. If you use a macro to dial out, like &dialout(${EXTEN}), the leg extension will became ~~~~s~~~~ and if it happens you transfer the call, that will be the callerid appearing on the other phone display. I am just rewriting all the dialplan getting rid of the macro and using gosub, even if asterisk is
2015 Jun 29
2
Product CDR/Queue/Meetme
...; *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] Product CDR/Queue/Meetme > > > > Hello, > > > > > > I am interested, too. > > > > > > Att, > > Welinghton > > > > > Citando Mitul Limbani <mitul at enterux.in>: > > Hey Helvio, > > Would like to check it out as well. > > Do email me, > > Mitul > > On 22-Jun-2015 9:05 AM, "Helvio Junior" <helvio.listas at gmail.com> wrote: > > Gentleman, > > Moderators, i don't know i...
2013 May 14
4
dial and bridge
Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a "channel" format, like SIp/1234 or Local/1234 at ext) and "park" it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. As a requirement, I cannot use the