asterisk groups
2005-Sep-08 04:12 UTC
[Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)
On Thu, 2005-09-08 at 16:26 +0200, Simone Cittadini wrote:> Is it true ? > My boss is just asking me if it is possible to stuck 4* TE411P in a > single server, for a total of 480 lines, someone can assure me it is > possible/impossible (manageable/unmanageable) from real-life experience ? >You might want to offload some of that PRI termination to an external device such as a Cisco AS53XX, Lucent MAX TNT, Audio Codes or Redfone fonebridge device and then trunk it to your Asterisk servers. But putting more then 2 quad cards in a single server is not safe. 1 per server would be more acceptable. This link might be helpful to you: http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large Good luck.
Wayne Gemmell
2005-Sep-08 06:47 UTC
[Asterisk-Users] Not enough lines available for Asterisk implemetation
Hi all I am looking at implementing asterisk at a company with two ISDN bricks (60 lines). I know that the VoIP will absorb at least on brick worth of lines but that still leaves me with a need for 30 ISDN lines. As far as I can tell most of the Digicom cards have 4 FXS ports and I've read on this list that at most two could coincide in a box simultaneously without causing an interupt flood. 1) is my info okay so far? 2)What would be the best way for be to implement the other 22 lines? Is there hardware I'm not aware of? -- Regards Wayne Gemmell Tel & Fax: (011) 894-4081 Cell : 072 836 4325 Email : wayne@terico.co.za
Simone Cittadini
2005-Sep-08 07:26 UTC
[Asterisk-Users] MAX PRI for single server (was: Not enough lines available for Asterisk implemetation)
> >that still leaves me with a need for 30 ISDN lines. As far as I can tell most >of the Digicom cards have 4 FXS ports and I've read on this list that at most >two could coincide in a box simultaneously without causing an interupt flood. > > >Is it true ? My boss is just asking me if it is possible to stuck 4* TE411P in a single server, for a total of 480 lines, someone can assure me it is possible/impossible (manageable/unmanageable) from real-life experience ? thanks *
John Daragon
2005-Sep-08 08:23 UTC
[Asterisk-Users] Not enough lines available for Asterisk implemetation
Wayne Gemmell wrote:> Hi all > > I am looking at implementing asterisk at a company with two ISDN bricks (60 > lines). I know that the VoIP will absorb at least on brick worth of lines but > that still leaves me with a need for 30 ISDN lines. As far as I can tell most > of the Digicom cards have 4 FXS ports and I've read on this list that at most > two could coincide in a box simultaneously without causing an interupt flood. > > 1) is my info okay so far? > 2)What would be the best way for be to implement the other 22 lines? Is there > hardware I'm not aware of?Wayne, hi; I guess what you're describing is 2 x ISDN30 connections (around 2Mbit/s each ?) I'm not familiar with the SA telephone system, but in the rest of the world (more or less) the card you'd need is the Digium TE110P which is switchable between T1 (24 channel) and E1 (30 channel) ISDN. jd -- John Daragon john@argv.co.uk argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
PistolPete
2005-Sep-08 08:37 UTC
[Asterisk-Users] MAX PRI for single server (was: Not enoughlines available for Asterisk implemetation)
We run many servers with 4 Quad cards and have no problems, SANGOMA works great for this !! -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk groups Sent: Thursday, September 08, 2005 6:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MAX PRI for single server (was: Not enoughlines available for Asterisk implemetation) On Thu, 2005-09-08 at 16:26 +0200, Simone Cittadini wrote:> Is it true ? > My boss is just asking me if it is possible to stuck 4* TE411P in a > single server, for a total of 480 lines, someone can assure me it is > possible/impossible (manageable/unmanageable) from real-life experience ? >You might want to offload some of that PRI termination to an external device such as a Cisco AS53XX, Lucent MAX TNT, Audio Codes or Redfone fonebridge device and then trunk it to your Asterisk servers. But putting more then 2 quad cards in a single server is not safe. 1 per server would be more acceptable. This link might be helpful to you: http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large Good luck. _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
steve@daviesfam.org
2005-Sep-08 22:51 UTC
[Asterisk-Users] Not enough lines available for Asterisk implemetation
On Thu, 8 Sep 2005, Wayne Gemmell wrote:> Hi all > > I am looking at implementing asterisk at a company with two ISDN bricks (60 > lines). I know that the VoIP will absorb at least on brick worth of lines but > that still leaves me with a need for 30 ISDN lines. As far as I can tell most > of the Digicom cards have 4 FXS ports and I've read on this list that at most > two could coincide in a box simultaneously without causing an interupt flood. > > 1) is my info okay so far? > 2)What would be the best way for be to implement the other 22 lines? Is there > hardware I'm not aware of?What do you mean by a brick? If your incoming lines are ISDN primary rates then you simply connect directly to a primary rate ISDN card in your Asterisk box. A single board from Digium can handle up to 4 primary rates. Give us a call on 021 6575160 or 011 6212270 and we can discuss further. Regards, Steve Davies
Did you search the maillist archives for hybrid echo cancellation?> Hello > > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) (everything > i > speak into SIP phone microphone i hear in its speaker). The person calling > from PSTN is not getting any echo. > > Which piece of the call could be causing the trouble so i can look into > it? > > thanks, > Marek