similar to: Asterisk Memory Requirements

I have a nice used zhone channel bank I want to experiment with, but need a T1 interface for my * box to do so. The TE410P looks nice, but more money than I want to spend to experiment, and I don't need 4xT1, only 1xT1. Are there any good 1xT1 PCI cards that are recommended? Regards, -Dorn

Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon

Below you will find, what I believe to be a typical setup with a T100P card. My question is - 1. Is this correct? 2. What kind of phones would be needed here... (Would you have to use Digital phones) And if so what would you recommend. PRI/T1----- | | | ---------------- | | | Channel Bank | | |

I just noticed that the new TE410 only supports 3.3v PCI. How do you know if it will work in your machine? It's not something commonly listed in your motherboard specs.... I was thinking about getting one for an Athlon 750 that is a spare box I'm doing testing on... (Original SlotA) -G

Hello List Just want to check if anybody else is having this problem. Every time the PRI connections are disconnected, the card freezes, and I have to reload the driver, to make it work again. We are very seriously considering switching to Sangoma at this moment, due to this and other problems, but I want to know if there is a solution, and to make sure it isn't asterisk that's freezing

I encountered this strange problem which is I can call into Asterisk box but I cannot call out. I was successful before using exactly the same euroISDN line but with TE110 and different versions of Asterisk. This time, I am using: . TE410 . Asterisk 1.4.13 . Zaptel 1.4.6 . Libpri 1.4.2 1) I put the following into extensions.conf to get to the outside line exten => 0,1,Dial(Zap/1) 2)

Hello, I don't have any experience with channel banks and would appreciate any feedback on my theory outlined below: We have a single T1 entering the building with channels 1-12 being voice lines and 13-24 being a 768k internet connection. This T1 terminates to an Adit 600 (T1-1). Here's what I know. Channels 11-12 go out the Adit 600's 25-pair connector to a wiring block (and

If I remove the eco cancellation module from a TE411P card, will it work as a plain TE410P? -- Carlos Chavez Prats Director de Tecnolog?a Telecomunicaciones Abiertas de M?xico S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 191 bytes Desc: This is a digitally signed

Anyone know who that good looking female is thats on the Digium.com website ? Ok, my Real question is I noticed that Digium has relesed a new T1 card with an echo canceller. I also noticed that its supports E&M Circuits. Im I have very little knowledge on T1 circuits and traditional PBX's so what Im asking is can I use Digiums T1 card to connect to another PBX via a tie line ? Or does

Hello, I would like to have some comparison between E1 cards from Digium and those from Eicon for a VOIP - ISDN Gateway. How does they compare on the echo cancel point of view? Is the echocancellation code for E400 good enough for production environment? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz -

This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78

We are located in Belgium and just ordered a PRA line, the telco asked the following questions : - 120 or 75 ohm ? - Support for CRC4 yes/no ? - B channels in 2 way ? We will buy a digium card but which one should we buy ? could anybody help me with this ? Thank you Michael Sorry for the previous html mail DISCLAIMER: The content of this e-mail message does not constitute a commitment

Hi, I'm running Samba AD DC in a VM under Proxmox. And it's eaten all RAM (1.8GB) within 3 days of running: [root at vm-dc3 var]# uptime 19:02:54 up 3 days, 5:04, 2 users, load average: 0.01, 0.05, 0.05 [root at vm-dc3 var]# free -m total used free shared buff/cache available Mem: 1752 1362 205 2 184

Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports.... Thanks, Isaac

We've just setup our A@H server, with our quad port card. Everything works well so far. One thing I notice is that when I leave the handset on the hook and dial a #, all is well. If I pick up the phone and dial, it cuts off at 10 digits, which is a problem if I need to dial 1+area+phone # (12 digits). The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a

Hello, I have a 2 bind nameservers on my network (both running C3.9). Over the last 3 weeks, I've seen a significant increase in the amount of denied cache queries to DNS service and I'm wondering if I should be worried or if there's something I can do to resolve or prevent this. I show millions of lines like: Oct 4 11:49:30 dns1 named[878]: client 68.13.16.20#53535: query (cache)

> dim(data) [1] 22283 19 > dm=dist(data, method = "euclidean", diag = FALSE, upper = FALSE, p = 2) Error: cannot allocate vector of size 1.8 Gb Hi Guys, thank you in advance for helping. :-D Recently I ran into the "cannot allocate vector of size 1.8GB" error. I am pretty sure this is not a hardware limitation because it happens no matter I ran the R code in a

I'm trying to solve a mystery for the last couple of days. I have a mix of D70, D50 and D40 behind NAT. Server is in a colocation, not a VPS. For several years, everything was working fine, no issues. A few days ago I started having problems at one particular site. NO CHANGES have been made to this office network - same router, switch and internet provider. No new equipment added or

Hi, We have a scenario where we receive calls from 2 different places: 1- Avaya IP Office 2- CIC Interactive Intelligence PBX and the calls are transfer automatically to an Asterisk Box. The problem we are experiencing is that more that half of those calls come with Echo and Jitter. For outbounds calls everything works perfect. I'm on RH9 Stable v1.4 Version, using an quadspad TMD400P digium

Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle