similar to: Switch from NBX to Asterisk

I sent this once already but it didnt show up in my mailbox so I am not sure if ever made it to the list. If it did, my apologies. I have an * system and an nbx. My plan is to use several grandstream 286s and plug the the phone cords into the NBX's analog FXO card. It works fine but after I make a call, the port stays open on the NBX and I cannot make another call for around five minutes

Hi, I've looked pretty wide on Google for this so I don't think it's been asked before. Has anyone had experience integrating Asterisk with a 3COM NBX system? The only way that I can see that looks possible is via the 3Com NBX ConneXtions H.323 product and then one of the H.323 Asterisk channels. Has anyone done this? Any pointers as to what to avoid or whether to even attempt it?

How hard is it to integrate * into an existing 3com NBX setup? -Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041004/bf01e3d9/attachment.htm

Has there been any luck getting the 3Com NBX series phones to work with Asterisk? Thanks! -Andrew

Does anybody know if the 3com NBX 100 phones will work with Asterisk? The phones require a boot image to be sent either through layer2 or layer3 before they will function properly after being powered on each time. Thanks.

There seems to be a problem related to the Grandstream HandyTone-286. When a call is placed through the adapter, the call can be transferred. However, when a call is received through the adapter, the call cannot be transferred. The problem does not exist with a BudgeTone-101 (1.0.4.23) using the same Asterisk configuration and Dial() settings (Ttm). I tried all of the firmware on their BETA

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Hello! Is there a phone compatibility list anywhere? I know Asterisk is supposed to be compatible with IP phones that support SIP, h.232, and IAX, but a list of phone known to be supported would be a nice addition to the documentation. I need a buisiness phone, and the Cisco is _EXPENSIVE_ I like the 3com 1102, but it is NBX phone, and I can find no documentation of this phone working

I am new to VOIP, Linux and Asterisk. Through a lot of reading (this list, voip-info.org, documentation, etc.), I successfully installed FC3 and * on a new Dell SC420 with two X100P connecting to two PSTN lines at my office. I've also installed AMP to help me configure IVRs, call groups, extensions, etc. I use a Handytone-286 ATA and x-lite clients on the internal network and all works

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

Hi All, I've an Asterisk Server behind a NAT. Using DNAT, I've opened port 5060 and all 10000:20000 udp. Sip configured with externalip and subnet. I've another site, also with NAT, where I map the rtp port (as defined in the client) to map to the local client (DNAT). Using Xlite, this configuration works, it requires using the quality=yes and NAT=yes/always in the sip ext

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

My new company is going to enter a bid competing with 3Com and Siemens solutions from others. Voice service is a PRI T-1 with a block of 100 DID numbers, a second T1 that will carry Integrated voice/data, 12 voice/768K for redundency. I asked this a week or two ago, now I have more details. Best/$ channel banks that work with Asterisk and Best deal on phones... 16 business, 34 basic IP

Hello, everyone, I am currently trying to get the asterisk server to talk with a 3COM NBX with T1 connection. My asterisk server has a T100p, TDM20B, a couple of sip phones. Now the sip phones are calling 3COM NBX phones fine, however, the analog phone has problem when dialing the NBX phones. The connection is established and the NBX auto-attendant picks up the call however the NBX end says that

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

Hi, I am quite new to asterisk so I am not sure what is needed to figure out this problem. If more information is needed and not provided I will gladly provide it. I have a very basic asterisk setup. 1 x100p card and a grandstream handytone 286. I can make calls fine to most phone numbers from the handytone device the trouble seems to come when I dial this number 591-1079. It puts me through to

Hello, I bought a Grandstream Handytone 486 to forward incoming calls from our old analogue PBX to the asterisk server. My first test was connecting an analogue phone to the Handytone and calling a sip phone - worked. Now I used the same cable to connect the line port of the Handytone to the analogue pbx. When I call the number of the analogue PBX I hear a clicking inside, but the call

Hi, I would like to simulate data with a binary outcome and a set of predictors that are correlated. I want to be able to fix the number of event (Y=1) vs. non-event (Y=0). Thus, I fix this and then simulate the predictors. I have 2 questions: 1. When the predictors are continuous, I can use mvrnorm(). However, if I have continuous, ordinal and binary predictors, I'm not sure how to simulate

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't