similar to: Micronet 5050s FXO gateway and hookflash transfers.

I tried setting up a Micronet 4port 2 fxs 2fxo device using sip. I found that I could get multiple fxs ports to register with *. I also found that no matter what dialed on as an fxs user when dialing out it would try to go out the fxo side of the micronet. I also had major problems with these devices behind nat. I was having so much fun with these little boxes that I put them back on the shelf

Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter >> I am located in the UK and

Hello, I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the problem is i can make call but can't receive calls. If i make a "sip show peers" it shows the micronet is not connected to the asterisk. Does anybody knows how to configure the micronet and asterisk to solve this problem ? Thank you

I think I've managed to figure out that there are two ways to transfer a Zap call, using hookflash (defined in zapata.conf) or the # key (the t and T options of the Dial command in the dialplan), but not why there are two ways to do this, nor what the difference is between them. Is there something that explains this? thanks -------------- next part -------------- An HTML attachment

Don't know what else to call this. Googling and some time on the IRC channel haven't gotten me anywhere. Here's the sitch, which is a bit complicated but is something my customers are in fact encountering on an everyday basis: 1. Bob is on a Zap channel talking through the PSTN to Carol. Both have the misfortune, like so many of us, of having LECs who do not offer disconnect

Hello people, I am using Asterisk with a handful of Micronet SP5100 IP Phones and a Micronet SP5052 FXO Gateway. So far I have incoming calls ringing all the phones correctly, outgoing calls working, voicemail working and calls between phones working. The only think I cant get working is Transferring of calls, and the ability to put calls on hold. The phones have both a 'Transfer'

I wonder if I'm the only one whose customers are having trouble with hookflash on their TDMXXX cards. The problematic situation of record for us is a user who is on a call, and then wants to do one of two things: Hang up that call and take another one coming in Hang up that call and make another new call What happens is that instead of seeing the event as a hangup, asterisk perceives

Is it possible to send an external hookflash command to the Digium FXO card from the asterisk PBX? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031007/fe5be94d/attachment.htm

The hardware we are planning to use is: Micronet SP5050 FXO Gateway http://www.micronet.com.tw/Products/VoIP/SP5050.asp Micronet SP5100 IP Phone http://www.micronet.com.tw/Products/VoIP/SP5100.asp We are hoping to use this hardware along with AsteriskPBX to replace our aging PBX system. What I want to acheive is: * Any incoming call from PSTN (via gateway) rings on the receptionists phone for

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

I'm sure dumb when it comes to describing things that happen on my system. I'm making an outbound call on my ATA186 when another call comes in. I first get the nasty CID screech and then the periodic call-waiting tone. So far, so good. Then I hookflash, and just like it's supposed to, the waiting caller is on the line. But during the duration of that conversation, my console

Hi Charles! Thanks a lot for your response. Now I figure why the isbmex driver was not working. On the other hand I've tried to run the command suggested. But no success. I got this information when running the dmesg command after running the modprobe $ sudo modprobe ftdi_sio vendor=0x0483 product=0x0035 [35816.111760] usbcore: registered new interface driver usbserial [35816.111792]

Hi all, Is there a way to change the hookflash timing with the TDM400P? Allready been searching the mailing list/google etc but i can't find anything ;-( I tried flash= in zapata.conf, but that only works with the T1/E1 cards. Greetz, Caspar -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using zaptel with three way calling and call transfers with a hookflash. If I try transfering a call to an extension that is engaged I get an engaged tone. This is fine, but how do I get back to the caller? If I hookflash again I seem to put on a three-way call and the caller can hear the beeping. I can press hookflash again but I'd prefer the caller to hear only the hold

On Feb 26, 2014, at 2:10 AM, Victor Romero wrote: > I got a UPS Sola Basic but it is not working, I'm providing the details. I hope that someone can help me to get it working. > > Please let me know if there is some other information that I should provide :) Thank you for providing all of this information! > OS name and version > Ubuntu 12.04 > $ uname -a > Linux

I got a UPS Sola Basic but it is not working, I'm providing the details. I hope that someone can help me to get it working. Please let me know if there is some other information that I should provide :) OS name and version Ubuntu 12.04 $ uname -a Linux olimpo 3.8.0-29-generic #42~precise1-Ubuntu SMP Wed Aug 14 16:19:23 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Exact NUT version, 2.7.1

My current config: pstn -> audiocodes fxo gateway -> asterisk -> xlite every fxo ports are registered with asterisk I have this extensions.conf exten => 111,1,answer exten => 111,n,dial(sip/fxo1) exten => 111,n,hangup If we dial 111 by xlite, I could hear pstn dialing tone. I could key in a phone no and connect to the called party. this is a two stage

Hi, every one I'd like to introduce some new feature of our products. mg3000-r fxo gateway provides more feature to work with asterisk. 1.play asterisk ivr with no interuption. when the mg3000-r received call from co line, it wouldn't conect instantly.instead, it start call to asterisk ivr first,when the ivr ready, it connect the co line. this feature make user feel friendly.

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

I just received this email in response to a question I asked of Qtelnet about their hardware's compatibility. I had specifically asked about their FreeRide FXO-1301 Gateway. Now, if someone would be so kind as to send me a demo unit, I'd be glad to test it :-) ----- Andrew Thompson ----- Original Message ----- From: "Matt Lukens" <email snipped> To: <email