similar to: SIP, * and clients behind NAT

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

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, 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 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

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

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ? Adri? Vidal -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 235 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/e5514052/attachment.bin

I have an analog Fax machine which I wish to connect to the network and the Asterisk server. It will connect through a GS Handytone 286 converter and then into the LAN. Is there any information out there on what I need to write in *sip.conf* and/or *extensions.conf* to make sure the fax works as a fax? Channel 8 on my T1 is a reserved, dedicated line for the fax number. Do I need to

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 ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

Hi Has anyone opened up a grandstream phone or handytone ATA to find out what is inside? What is the CPU? How much RAM? Cheers Rob

I have used the handytone 488 from grandstream in the past.... However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. Jerry

Hi I have a analog phone connected to my Gateway Handytone and registered to Asterisk 1.4 I have configured my HandyTone 488 (in the section FXS Port) for make and receive calls, however I can not transfer a call when it come via PSTN. But, when a call come from via IP I can transfer it. [phoneanalog] type=friend secret=XXXXXXX context=local nat=no qualify=yes host=dynamic dtmfmode=rfc2833

Hi anyone know if i can connect a Grandstream HandyTone 488 to Asterisk ? Actually my HandyTone 488 are connected to: wan port to my lan line FXO port are connected to my local analogic line i want that when a call in by my analog line, it's sent to my asterisk for other voip post can answer .. it's possible ? thanks bye

Hi everybody! I'm from Argentina, so you'll have to sorry me for my English. I have a Linux box with asterisk and want to buy an ATA. Fist, I thought about the Grandstream HandyTone but I read some reviews which says that it has a lot of echo. Some people recommended me Sipura 2000 but I don't know what to do. Now I just to make some tests at home and see what happens and if it works

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

Guys, Im new to asterisk and voip but Im have a couple of questions regarding the initial setup. 1. Im going to install an asterisk server at home, where I have 2 phone lines, what kind of card do I need to get? I was thinking about 2 X100P Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? Ive seen the Grandstream HandyTone HT-286, I guess that servers as and FXS devide

Hi; I've just had a couple of discoveries in the learning process that are really making me impressed with this software. 1) IAX transfer - I'm running Asterisk boxes A, B, and C. B is in the middle and has a dialplan that points to extensions on C. When a client on A in the proper context on B tries dialling a client of C, B is smart enough to release itself somehow from playing

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a