search for: a108d

Displaying 10 results from an estimated 10 matches for "a108d".

Did you mean: a104d
2009 Jun 24
0
problem with sangoma card a108d
Hello, I need help to use my sangoma card a108d. I need that another server give me an E1 with a clock. The server with the sangoma reseive the E1 clock on port1 and is MASTER E1 on port2. But, I cant receive the clock (I am connected). Anyone can help me? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnemen...
2007 Sep 24
2
Sangoma or digium ?
Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij leon at scarlet-internet.nl
2009 Oct 20
3
High Volume Call Center SIP versus IAX2
I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to make sure I am approaching the setup correctly. Thanks -------------- next part -------------- An HTML at...
2005 Dec 16
8
HW Echo Cancellers
Hi, To solve echo problems, I'm considering 2 alternatives. 1> Sangoma A104d - I can't find support for asterisk 1.2.1 2> Desktop echo canceller - http://www.oriontelecom.com/echo_canceller/desktop/e1_ec_desktop.html - I want to know where to buy and price. Any suggestion is appreciated. Thanks. Jason. p.s. : asterisk cli command "reload" can change rx_gain and
2010 Oct 01
2
No translator path exists for channel type DAHDI (native 76) to 256
Hello, We are having issues with a NEW Sangoma A108D: -- Executing [691918892 at pbx1:1] Dial("SIP/xtravoip200-009d24b0", "DAHDI/g0/691918892|30|m") in new stack [Oct 1 10:04:43] WARNING[14171]: channel.c:3170 ast_request: No translator path exists for channel type DAHDI (native 76) to 256 [Oct 1 10:04:43] WARNING[14171]: a...
2007 Mar 30
1
One way intermittent static to PSTN
...ving intermittent problems where the person we call reports static when we place an outgoing PSTN call. Only the person called hears static, to us the conversation sounds fine. Never happens on inbound calls. It doesn't matter what channel you call from (IAX, SIP, or Zap). We have a Sangoma A108D with hardware echo cancellation with 2 PRIs to Level3 and 2 PRIs to a Nortel Option 61c and then several IAX trunks running into this box as well. Box is HP DL 385 G2. I've ruled out bad cables, bad port on sangoma and bad port at Level3 rack. When under load and while the problem is occurri...
2006 May 29
8
E1 hardware for asterisk
Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier
2010 Oct 21
1
Hardware Compatibility HP Proliant - Sangoma PCI Express
Hi to all, I am in the process of setup a new asterisk server, I think in the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE Card. The specs of the Proliant (HP PART 487932-001) about PCI are the next. 1 ( 1 ) x PCI Express 2.0 x16 ( x8 mode ) , 1 ( 1 ) x PCI Express 2.0 x8 ( x8 mode ) , 4 ( 3 ) x PCI Express 2.0 x8 ( x4 mode ) The question is, if
2010 Dec 10
0
HP DL360G7 with T1 card(s)
...the USB subsystem like on the recent DL120 boxes? I'd be interested in the output of: dmesg | grep 8042 2. Assuming the PS/2 ports are real, did you need to disable the USB subsystem in the BIOS to get good performance from the T1 card? 3. Has anyone used one with the 8-port Sangoma A108E or A108DE? 4. Any gotchas found by anyone? Thanks Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org
2009 Nov 11
2
Bug or feature: SIP chanvars not overriden
Hello, Using 1.6.2-rc5, my settings include: [local-phone](!) context=mylocal type=friend nat=no canreinvite=no host=dynamic qualify=yes dtmf=info language=fr call-limit=5 subscribecontext=subs disallow=all allow=alaw t38pt_udptl=no setvar=accountcode=foo [168](local-phone) defaultuser=168 secret=pass168 callerid=John Doe<168>