search for: handyton

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 exce...

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...

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 b...

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 kullan...

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 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 qua...

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 the local 911 dispatch. Causing the police to show up at my doorstep and check to make sure everything is alright. I can see why I think; 5...

...ut 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 alternative to the PAP2T, and I'm seeing prices hovering between $25 and $30. I'm considering getting one of these Grandstream ATAs onsite to play with before I make my final decision. What do people think about both products? Bonus points for if peo...

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...

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 create a special group in *zapata.con...

Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running Asterisk) and have configured a catchall extension to recei...

For some reason, this didn't go through the first time, maybe because I had JUST signed up. Hello, I'm trying to configure Asterisk and my Handytone 488 to pass incoming calls coming over PSTN through the FXO port to Asterisk, which will process the calls with voicemail, or some such service. I point the Handytone 488 FXO port configuration to 192.168.0.2 (the machine that is running Asterisk) and have configured a catchall extension to recei...

can someone who has a grandstream handytone 488 working with making outgoing calls through the fxo port please post the parts of their config that deal with this port? i cant quite seem to get it to make outgoing calls despite having tried several completely different ways of making that happen. i have been told that asterisk@home has this...

Hi, Yesterday I finaly got my handytone sip adaptor. It works.... But when dialing to and from ISDN I got echo in both ends, I had tried diff. codecs, but then the GS wont work at all - It can do a call, but after 3 'ring' it disconnect. Any hints ? _________________________________________________________________ Scope out...

...de 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 canreinvite=yes dtmfmode=info incominglimit=1 disallow=all allow=ulaw allow=alaw [4008] ; HandyTone 286 with POTS phone type=friend insecure=yes context=test-budget username=4008 fromuser=4008 callerid=4008 host=dynamic nat=yes canreinvite=yes dtmfmode=info incominglimit=1 disallow=all allow=ulaw allow=alaw I can send the history file - but it's quite large... Thanks in advance... ...Paul

...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 configuration but works quite well. However, lately I've purchased a Grandstream ATA Handytone 286 and tried to apply the same settings but... When doing an echo test, I can't hear myself, but I can hear the asterisk server (meaning asterisk can reach the client behind the NAT). When doing some tcpdump, it looks like some packets are coming from the client to asterisk, so the netwo...

I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have

Could anybody shed some light in which device they would use in this situation: Remote office PBX's to be connected via a) Cisco ATA-186 or b) Sipura SPA-2000 or c) Grandstream HT-ATA-286 to go via the net to an * box. Pros / Cons for each device would be appreciated! Thanks Kris

Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences?

...ux 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 fine. I would like to connect to * as an extension from home, from client sites, from hotels, etc. Most of these places will be behind some type of NAT and/or firewall. At my home, for example, I have a consumer grade firewall/NAT...