similar to: how scalable is digium cards?

Hi all, I'm confused, in digium website, it says: TDM2400P: It supports a combination of up to 6 FXS and/or FXO modules for a total of 24 lines. 6 plus 6 is 12, how come it's 24? if I have 24 PSTN lines, i'll be needing 24 FXOs. Pls. elaborate. thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, is it possible to define a pilot number in asterisk, say I have 3 direct lines and I want one of those direct lines to be used as pilot number? When that number is contacted it will be redirected to the available zap and original zap that receive it will be freed to receive another call. It can only be used when all 2 lines ares used. Thanks. Lito -------------- next part

Though I've had implementations of Asterisk, I havent encountered this one yet, so i'd like to seek your advise if this possible. I would want asterisk to be stand between the dialer the destination. The dialer will now dial asterisk access number. Asterisk will acknowledge user by using CallerID and check against its database for authentication and then sends out a DTMF A tone for ?

> Lito Lampitoc wrote: > > thanks for enlightening. So you mean, if I have 3 lines when the > caller > > dialled the first line and it was busy, the call will be diverted > to the > > next two available lines in random? > > > > I don't think it's random. I think its just sequential. If main > line > is busy, try second. If that is

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

I've seen this consistently on three systems, with three different cards, and multiple versions of DAHDI. At first I thought the issue only occurred on newer, Nehalem-based, systems, but I reproduced it on a Core 2 Duo box as well. I've tested with dahdi-linux 2.2.0.2, dadhi- linux-complete 2.0.0+2.0.0, 2.1.0.2+2.1.0.2, and 2.2.0.2+2.2.0. The card is a Digium TE220B which uses the

Hi all, Does anyone run Asterisk on FC4? with Digium's TDM40B cards. any success stories? my Intel 865 M'd+ Intel 3.0GHz freezee during installation (FC4). Please any comments? Kumara

Hi All, I Have a Digium Card TDM40P, the specification say: OEM TDM40B: TDM400P + 4 PORT FXS Bundle, my question is: Can I to install analog lines of PTSN?, other detail is: this card have 4 card green. I need to know what is the best card for the following scenario: I need a IVR for my comapny and a PBX, but i want that my extension not use FXS I want IP phone . Thanks ins advanced,

Hi, I'm looking into support for SS7 and I found an article (http://www.openss7.com/news13022002.html) which says that OpenSS7 supports Linux Support Inc's T400P-SS7 Quad T1 and E400P-SS7 Quad E1 PCI interface cards. It also says that Linux Support Inc is the primary sponsor of Asterisk. However I cannot find these cards on the Asterisk hardware page

Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the

Hello to all I'm using asterisk 1.4 and dahdi. I had everything working fine, and I could place calls through my R2 channel. But now the channel is always "RED" and Im getting this error message: TE4XXP: Version Synchronization Error! Here is my chan_dahdi.conf------------------------------ [channels] language=en context=incomingr2 signalling=mfcr2 mfcr2_variant=ar

Just curious... Does anyone know if any companies using Asterisk on embedded hardware (out at the customer premisis), such as the Soekris Net4801, to provide VOIP service? Doug.

Hi I got the 2nd Gen firmware upgraded on the TE405P. I recompiled after putting in the upgraded board but did not change any conf, but the spans become active but will not come up. I guess I am missing something or are the any changes to the zaptel/libpri software that is required. I cannot find any info about this or does this new firmware only work with latest CVS. I am using 1.0.9 with

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

Hello I have an office with a T1 that provides 4 (out of 8) analog PSTN lines thru an adtran board. I want to add 4 more analog lines. Currently I have a Digium TDM40B. I'm wondering what the best upgrade path is, where I define 'best' as the solution that is most likely to work without problems (like interupt conflicts) and work with my current echo tuning . I see my purchase

Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [root at catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/ RED 1 PRI CAS RED 2 PRI CAS RED 3 PRI CAS RED 4 PRI

Hi all, Which is the practical (from your experience) limit of the number of X100P cards installed in a single Asterisk box? Asterisk can work reliable with 6 X100P cards in the same box? Anyone know when the 4 ports FXO Digium card will be available on the market? Many thanks, Dan P.S. Please do not aswer with RTFG ...tried before without success...:-))

On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does the processor have? We're now seeing chunks of missing audio and I can't tell whether this is due to a kernel upgrade or to a zaptel/libpri/asterisk upgrade. I'm not seeing missed interrupts (from a cat of the proc/zaptel files), any other ideas on how I could go about tracking this down? I'm

I went on a service call yesterday to find an asterisk system with a T100P card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B LEDs were not lit, and the system did not respond to keyboard input. However, calls were being processed for the PRI and 7960 phones. I replaced the TDM40B card with a new one, and the system now seems to be ok. But I'm wondering, why