similar to: SOLVED - Channel bank woes - no outbound calls

So, I'm still having this problem with outbound calls not working when using a channel bank. I've purchased a Rhino FXO channel bank from VoIPSupply.com to make sure it wasn't an equipment problem. I am using a Digium TE411P card, and have simplified it down to just 1 port plugged into the channel bank, with just 1 analog line plugged in. If I place an inbound call on the line, it

Hello list, I have recently deployed Asterisk as the phone system for my office. So far, everything has been going really well, except for one little thorn in my side. I have a set of 6 analog lines that are connected to a TE411P via a Rhino FXO Channel bank. If I call the analog number, Asterisk answers the call, and routes it appropriately. The problem is, when I hangup, Asterisk never

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

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

Hello all. I'm having a problem debugging an IVR I'm building. I can't see any reason this shouldn't be working. Firstly the asterisk version is: Asterisk SVN-trunk-r7230 built by root @ localhost.localdomain on a i686 running Linux on 2006-02-17 22:44:48 UTC Basically the problem is this. While the playbacks are happening you can push any one of the options and to happily

Anyone ever experience noise instead of a dial tone. We are using a Digium TE410P and 2 units of Rhino FXO & FXS Channel Bank. Everything works well when they are freshly turned on or rebooted, but after a while there's only noise can be heard instead of dial tone. Does this have anything to do with the timing configuration? Where is this time is generated from, is it from the TE410P card?

Has anyone had any success with the Rhino CB-24? I can't get mine to work, I tried all the obvious settings. The cb-24 gets stuck at init ESF framing, as if it's not seeing the t1 card at all. It does get a t1 carrier (detecting voltage??) Help! Thanks.. Everything appears to look good on the server, I did: [zaptel.conf] span=1,0,0,esf,b8zs fxogs=1-24 [zapata.conf] signalling=fxo_gs

I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=2,0,0,d4,ami fxsks=25 And in zapata.conf, I

Hi, I've dug a bit through the wiki and the mailing lists, and haven't really seen anything like this, but there must be someone out there doing this. Basically, there is a fax line that I don't want to answer inbound, but I want it available to do dial out from. Right now, we are using a busy wait around the ringing line, but I was hoping for something that might be a little more

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Hi all I'm trying to configure a TE410P in Europe with three E1s and a T1 channel bank (already bought in the US) to receive and make calls through *. I managed to get the E1s to work, but I'm having trouble with the channel bank (a Rhino). I have tried the bank in spans 1 and 4 changing jumpers in the card (always with no PRI connected, as I have no PRI where I'm testing now,

I am forklifting a Merridian option 51c with 112 Nortel Digital Handsets and 400 analog units. For the analog units I have quotes for 9 ADIT 600 48 port fxs units and 17 Rhino 24 port FXS channel banks. I have used neither. Which is the best choice? The price difference is not that great. I am looking at Citelinks 24 port Handset Gateway for the Nortel Digital units. (Any other suggestions

Hello all, I have a new Carrier Access Adit 600 and am having a problem dialing out and receiving incoming calls. After the initial ring, the calling party will here nothing. There is a silence, but the SIP phones that is supposed to ring will ring. The calling party will not hear the sound of the ring that * sends out on the line. Additionally, if someone picks up the receiver there is no

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

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.

I would like to have your comments regarding: 1) drivers/rhino.c ------------------ The following patch addresses some problems in drivers/rhino.c. Or that is what I think. I don't use this hardware. The patch: 1. Eliminates some usefulness definitions. 2. It removes code that evaluate the time. I can't see how the code actually does useful work. 3. The

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

I've read several places that say you cannot have more than two 4 port Digium T1/E1 cards, as it would overload the PCI bus. Is that true? If not, what is the maximum number of cards people are putting into boxes? If two cards is the limit, am I right in understanding that the preferred way to support a large number of users is to split out into multiple Asterisk servers, and then use IAX

I'm running the samba-client-3.0.20-0.1 SUSE RPM. I was using the version that came with 9.3 but upgraded to see if this specific problem would go away. Guest access does not appear to be working correctly, and it looks like the problem is due to guest not getting mapped into the LDAP query correctly. Specifically, I can login with local account, join workstation to the domain, browse