similar to: Adtran power consumption

-----Original Message----- From: Chad Brown Sent: Tuesday, December 21, 2004 8:02 PM To: 'el_flynn@lanvik-icu.com' Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not thePSTN Flynn, Yes, that makes sense. However, in my case I have incoming calls arriving on an IAX channel from a PSTN gateway. I think the concept is the same. That said, if incoming calls have access

Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same

hi all, i've got a proposed setup that i was wondering if you guys could comment on. the client wants * and a couple of SIP phones to be on a separate network than the rest of the office, so that in case their primary network crashes for some reason the PBX won't be affected. one other factor: the client may at some later point set up SIP UAs sitting on the primary network that will

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

I just started poking around with writing a python module to interface to the Manager API, and it suddenly hit me... how the heck are you supposed to program this thing? All the events seem to be dumped to all the open connections. If I send a command, such as a login, there seems to me to be no way to determine which response are intended for me, and which may be intended for another open

Hi all, I was wondering if anyone had any pointers on how to determine whether or not a module has gone wonky on the TDM400P? I have a 2 FXO (channels 3 and 4) and 2 FXS unit (channels 1 and 2). The bad (?) module in question is the FXO module on channel 3. I can't dial in to or out of that channel; dialing in gives a busy signal, dialing out just shows * hanging around after attempting a

I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at http://sig.lange.googlepages.com/assman . The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been

I thought this might interest a few asterisk users. I don't use them so I have no idea about Adtran's but I know a heap of people on this list swear by them. Cheers, Dean ________________________________ From: vclass.coordinator@adtran.com [mailto:vclass.coordinator_adtran.com@mail.vresp.com] Sent: Monday, 19 September 2005 9:59 AM Subject: ADTRAN Virtual Classes: Ensuring

Rich, Thanks a bunch, totally understand now and that actually makes total sense. (no need for schematics). This also explains why I used an TA750 to go into a Nortel MICS system, using FXO and no buzz. Totally balanced load from the analog ports on the Nortel across the 5 feet of CAT5 to the FXO on the adtran. Now I need to get rid of some Adtrans --- Anyone lookin to buy? :) Thanks

Hello, I have just received Adtran TSU 600 with 24 FXS ports. I have installed sucessfuly T100P card. Adtran is connected to t100p with crossover T1 cable. On T100P card I have a green light and on Adtran I do not get any errors or alarms. But I do not get dialtone on FXS ports. Adtran is configured: For Network Timing, fxs ports ore fxs_ls on Adtran. In zaptel.conf: span=1,1,0,esf,b8zs

I believe it is not fiber, but I am not sure. I am going to take one of them home tonight and hook it to my POTS line there, which for sure is not fiber. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Welter Sent:

I am sending a call down a E&M wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=>799179,1,Dial(zap/g2,20,D(9179)) exten=>799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered.

I have following setup Asterisk - T100P -> Adtran TSU600 P + FXOcard -> PSTN line When PSTN line is plugged directly in to analog X100P caller id is received by Asterisk but when I plug it into adtran I'm not getting caller id. Any ideas what kind of setup Adtran TSU600 requires to pass caller id to T100P ??? regards m.

I've never set up an mgcp device before. I have an Adtran IAD with the MGCP firmware on it. I have it configured in mgcp.conf like this: [general] port = 2427 bindaddr = 0.0.0.0 [adtran] host = 192.168.2.2 context = default canreinvite = no line => aaln/1 line => aaln/2 The device is configured like this: MGCP Configuration | Standard MGCP 0.1 / NCS 1.0 MGCP Endpoint

I just wanted to ask about using Adtran boxes to support analog lines into an Asterisk box. Right now x101p's are just too sensitive to RF noise inside the PC. Going with an external chassis looks like a good, albeit expensive option. It looks like I can use the Digium T1 card into an Adtran 850. The Adtran 850 would need FXO cards. It would also need the echo cancellation module. Is

Hello Can somebody send me please config files of zaptel.conf and zapata.conf for adtran fxs ports. I cannot make it work. I do not get a dial tone on Adtran and when I am trying to call from sip i get: app_dial.c:674 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time There are no errors on T100P and adtran. Please help! B.

Greetings, We have An Adtran 616 Total Access device talking to a colocated Asterisk machine over MGCP. Calls placed to the phones connected to the Adtran go through as do outgoing calls from the phone (prefixed by 9), but feature access codes (*97 for voicemail, for example) and extension-to-extension calls don't work. As soon as the first digit is pressed, the user hears a busy signal.