similar to: Grandstream 488 - VoIP-to-PSTN Calls

Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:

Has anyone tested a grandstream 488 FXO gateway on an Asterisk machine? I read that the 488 has a FXO port on it, can I use the grandstream 488 to pass traffic to the pstn from Asterisk. I would use this at home to pass traffic into a foreign country's PSTN over the internet. Thanks.

Hi, spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2), but rxfax doesn't work. After some FAX sounds, it hangup! Could someone tell me how to debug? The following is the * CLI> log to 192.168.0.161:43222 -- Executing NoOp("SIP/4881-bde9", "") in new stack -- Executing RxFAX("SIP/4881-bde9",

Hi all, Im having problems using a fax machine conected trough a grandstream 286 sip ATA, it must be able to send and recive fax from pstn, but fax always ends with communication errors 252/244/232 and others. Im using alaw/ulaw codes on pass trough mode, also have tried asterisk faxdetection, nvfaxdetect, disable echo cancellation by hand always with same results. Grandstream ATA is using

Hi, I have HT488 with it's FXO connected to Israeli PSTN (bezeq) when dialing to that PSTN line asterisk see gets the call and direct it to the right extension but if the extension doesn't answer and the dialer is hanging the call the extension will keep on ringing. I'm not an expert but it seems like my asterisk doesn't recognize the hangup signal from the HT488 -or it's the

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

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 built in to just a button hit, but i dont want to

Hi All, I have a GSM box, which needs to connect to a analogue phone line. I've plugged the GSM box to a Grandstream ATA (386). This ATA has extension number 600. Now what I want to accomplish is the following: - If a mobile-number is chosen by a user, asterisk needs to call the ATA (600), wait for a few seconds, and then send the mobile-phonenumber. Or, if it's possible, define the

Up until yesterday I've had a lot of high pitched noise when connecting a BT101 to the PSTN via the X100P. I was using an Asus A7V133 with raid motherboard and an 850 AMD Duron. Over the weekend I thought I'd try another machine. I had an HP Vectra 400 Mhz PII MMX with 128 Mb RAM available, today no noise at all. Now I must see if the Vectra is up to the job. -- Dave Cotton

Sorry if this question already asked. I'm implementing Voip with asterisk and grandstream gxw4108, according from the manual, for connecting with PSTN I must configure one SIP account and assign that for dialing the PSTN so in my sip.conf I configure SIP account(extension) : [1401] type=friend username=1401 secret=1401 host=dynamic context=my-office insecure=port in my extension.con

Has anyone used either the 8 port or 4 port FXO device from Grandstream? (GXW-4108 or 4104). They seem to be the lowest cost multi port FXO devices that I can find, so I'm getting ready to buy the 8 port version. I just want to see if there are any opinions on the device before I commit to the purchase. If people have not used the Grandstream, are there any issues with using

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 receive the call: [from-pstn] exten =>

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

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

Hello..I have a Grandstream 4104 (4 FXO) gateway connected to an Asterisk server and a traditional PBX..Asterisk users are able to call the PBX users but PBX users dont have access in Asterisk..Does anyone know if specific configurations in Asterisk and in Grandstream have to be done? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have a question on VoIP adapters. As far as I understand, those adapters are usually used to connect DSL/Cable access to a normal phone (Internet to Adapter, then to PSTN phones). I want to know if you can use those adapters to do the opposite: connect a few lines (1-4 let`s say) to the adapters, then deliver via SIP to an Asterisk box. (I know I could use a TDM400 and Asterisk, but I

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

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?

I have the following: Call -> PSTN -> * -> GrandStream 101 1.0.3.81 The GS displays "ohn ro n2600" when the call is past to the GS. If I pass the call to a XTEN client, Caller ID shows up. Any ideas ??

Hi, Hoping someone has run into the same issue. I have an * 1.0.5 tdm400p and 2 fax machines on grandstream 486 boxes. When a fax comes in, no problem receives it fine. When you try to send a fax out just as the fax seems to be finishing the send you get a comms error on the fax machine and it fails wanting to retry (tried 2 different brand fax machines same issue). The 486's were