similar to: FXS channel banks

I have a user who's system crashes on pri hangup request. Tried 1.4.19.1 and 1.4.20 as well as the latest libpri no change Progress is as follows...... < Supervisory frame: < SAPI: 00 C/R: 0 EA: 0 < TEI: 000 EA: 1 < Zero: 0 S: 0 01: 1 [ RR (receive ready) ] < N(R): 025 P/F: 1 < 0 bytes of data -- ACKing all packets from 24 to (but not including) 25 -- Since

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 found this today, I am not a skype user but have read on chan_skype and don't like aspects of how it is implemented. My thoughts on it are only theoretical as I haven't used it I just cringe at adding X to a server. Anyhow there is a new project called sippyskype that appears to do a similar sort of thing with a couple differences. 1. Its

Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4

Hello! I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs. In this configuration, we have trouble maintaining stability. It may be fine

Hello, I would like to consulate with you guys. I'm setting up an Asterisk server on Debian. The problem is that Rhino drivers are only compatible with Zaptel 1.2. By default debian stable offers asterisk 1.2 and zaptel 1.2, and that suits our needs. Is there a bleeding need to use latest version of asterisk? I have managed to install Asterisk 1.4 and Zaptel 1.2 but then i got the

I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not

Hi all... I did some Google searches and didn't find any info on this so I'm posting it here... if this was recently discussed, I apologize for the duplication -- please point me to the appropriate thread. System Description: Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard Intel Pentium 4 2.8 Ghz CPU 2 GB DDR2 Memory Digium T400P 4 Port T1 Card CentOS 5.1 (Final) Kernel:

I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI

Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a "graceful restart"). Besides, would a "restart gracefully" actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again

Two questions that was not always clear for me [sorry for posting to this list :\]: ############################################################################################## Q1) when cabling, is the color order important? like: straight cabling: A side: white-orange, orange, white-green, blue, white-blue, green, white-brown, brown B side: white-orange, orange, white-green, blue,

Hi guys, I'm new to the telco game and still pretty new to Asterisk, although I've been using it for a couple of months now and like most of what I see. At my office, we've got a small two extension setup w/ two Digium cards for a single FXO line and three FXS extensions, but I'm also working on designing a larger installation for a customer which will involve ~16 analog

Hello all, I am trying to connect asterisk to a Siemens Hipath 3750 PBX system. I have a physical connection issue. I know that I should use a crossover RJ48 cable to link the two systems. The problem however is that the physical interface of the Siemens system is very unfamiliar. From my digging around, I think that this is an S2M interface. http://www.mail-archive.com/asterisk-users at

Anyone have any experience with the Rhino T1 channel bank? It looks very cost effective at around $1300 for 24 lines but I haven't seen it mentioned on the Asterisk list yet.

Hello, Again one more question about hardware. What could you suggest me to buy. I need hardware to connect let's say 4 analog lines. (FXO). This hardware should "talk" to Asterisk of course.. Thanks very much for some advices :) Bartek -------------- next part -------------- An HTML attachment was scrubbed... URL:

More than 60% of our outbound calls are now to mobiles, so the time has come to whack in a gsm channel bank. Does anyone have any preference of bank ? Do you use a PRI or VOIP connection from the bank to asterisk ? Real-world experiences are sooooo much better than marketing blurb ;) We currently have a TE412P with a free socket, so we have a choice either way. I am looking for up to 30

Hi, Digium stopped to produce TDM400P and the new TDM410 is too new to find it in our shops. The only alternative available is a "fully-compatible" Openvox product...but is it really "fully-compatible"? Any experience about Openvox products (card and zaptel versions, etc...)? Thank you! Giorgio.

hi, we are currently planning are large site which will migrate from an old siemens hicom pbx to asterisk. it will be a slow migration, the asterisk server will be inserted between the telco E1 and the hicom. new phones will be sip ones. the customer has several fax machines and analog phones (some of them have to be explosion-proof). around 50 analog ports in total are needed. as we are in

Hi, Not sure if anyone will be intersted but I have put together an install guide for Asterisk on RedHat 9.. Its nothing special but it may be of use to the newbies.. Like me.. If you would like a copy let me know and I will send it to you.. later.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

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Hello All, I'm looking to get some feedback on which solution of providing FXS is going to have the best results with faxing. I'm only looking to see what method is going to provide the best digitization into Asterisk, not for transmission from Asterisk to else where. Any recommendations of specific channel banks are welcome. I will need to provide approximatly 216 FXS Ports and need