similar to: asterisk-users Digest, Vol 104, Issue 53

Hello everyone. I am looking for a E1 PRI card which supports network side signaling not CPE. The main idea is to connect an plain old E1 compliant PBX which doesn't have an VoIP module to the newly created VoIP infrastructure. Could we use a Digium TE122P or something other to resolve this situation? Thanks in advance. Dimitar -------------- next part -------------- An HTML attachment was

READ READ READ .... http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22,

With that kind of load, your users shall start complaining about choppy audio or voice clarity on random occasions, and you wont have a clue where to look for the problem. Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in

Hello Doug, Maybe you can have it uploaded on GitHub.com as a repository ? With a README.md file on how to install it for PHP7 ? Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On

1.8 or higher. Att, H?lvio Junior SafeId - Gest?o de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.junior at safetrend.com.br On 29/06/2015 14:43, Abdul Basit wrote: > Hi Helviom > > I am interested to evaluate your product. > > What asterisk version you build this product around? > > -- > regards, > > abdul basit | p: +92 32 1416 4196 | o:

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

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/20160217/0d4c2800/attachment.html>

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

Please keep the ?me to? emails off the list. Regards; JV From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Magno Guimar?es Sent: Monday, June 22, 2015 3:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Product CDR/Queue/Meetme Hello, I am interested, too. Att,

Hello everyone. I want to migrate an old PBX which uses the ?1-PRI from one Telecom to VoIP by transparently moving the numbers one by one. I mean that the numbers that the PBX handle must be transparently moved from one operator to another. The old connection to the PBX is ?1-PRI and we must preserve that because no one knows how to configure this PBX. So my idea is to connect a PC with 2

I am running the following rsync command to synchronize directories between two servers: rsync -axvz --delete-after -e ssh /SRCDIR/ blabla at DEST:/DESTDIR The transfer starts and after a short while it appears to hang after some files have been transferred. The process establishes connection on both sides so I did an strace from the remote machine (using the rsync-debug script as described in

Thanks Harry. I will check and revert. I hope it work perfectly with asterisk. Regards On Wed, Feb 17, 2016 at 8:32 AM, Harry McGregor <hmcgregor at biggeeks.org> wrote: > Hi, > > For analog, I really like telco grade channel banks. > > I would recommend the adit 600, there is a good market on Ebay, and you > can do 48 channels per adit 600, with 2 T1 interfaces. Having

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 if this topic

Hi Helviom I am interested to evaluate your product. What asterisk version you build this product around? -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 On Tue, Jun 23, 2015 at 7:34 PM, Tech Support <asterisk at voipbusiness.us> wrote: > Please keep the ?me to? emails off the list. > > Regards; > > JV > > > > *From:*

Snom default configuration is SRTP enabled. You should disable the SRTP from the phone web GUI configuration Sincerely, Sam Basan From: Mitul Limbani [mailto:mitul at enterux.in] Sent: Thursday, November 12, 2015 5:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] No sound with internal

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

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this? Neeraj Chand Support Analyst Fiji Islands Australia T: +6793342526 T: +61388924326 M:+6799344012 New Zealand www.ocis.com.au T: +649 980 7022 -----Original Message----- From: asterisk-users-bounces at

AFAIK g729 patent is expiring sometime in 2019-2020. Mitul Limbani On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigoseth at gmail.com> wrote: > Hi Steve, > > I understand your question and your point, but I use the g729 codec from > the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 > without a single problem. > > So, sory but I

Hai, Test again but now install 4.4.2 >> All versions below 4.4.2 are affected i.m.o. << My guess, bad patch somewhere, I do think security-2016-04-12-prerequisite-v4-4-regression-fixes.metze01.txt is the problem, since this is the only patch i remove from the debian patches. (compaired to the samba 4.4.2 from source and the 4.4.1 from debian) (!! Above works for me, i dont

Hi list, Some more useful information - I compiled 4.2.11 on CentOS 7.2, and tested with the same configuration files and AD DC, running on MS Windows 2012 R2. The issue still remains. Regards, Dimitar > From: dimitar.hristov at outlook.com > To: amartin at xes-inc.com > Date: Wed, 20 Apr 2016 13:23:05 +0200 > CC: samba at lists.samba.org > Subject: Re: [Samba] wbinfo -u,