similar to: Grandstream Budgettone 100 & 102

hello, i found in list archives some notes about grandstream sip voip phones. Does anybody succesfuly tested those phones with asterisk ? Mark ? What about the prices ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ ------------------------------------------------------------ A mind

[Posted here becasue your mail server is rejecting my direct reply to you.] Hi Martin, AFAIK SIP can run on both UDP and TCP but I have only seen it used over UDP.. :) To setup the GS phones you need to open up the following ports (If its still set at the defaults)... UDP/5060 UDP/5004 UDP/5005 UDP/5006 UDP/5007 I have not tested the GS phone through a firewall yet but this config should

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry

Does anyone know of any reliable supplier for Grandstream phones? I tried dealing with David Li from Grandstream, but after emailing him an order in August, and asking how he wanted payment, I never got a reply... James Ho from DGTimes was happy to give me pricing, but when I sent him an email asking for shipping costs, I never got a reply... I tried dealing with John from Chagres Ventures, but

--- "Chandra" <chandra@digital.com.np> wrote: >i have been using grandstream budgettone IP phones and they work fine >except that these phones times out after some hours.. i ahve seen that >the phones working ok are next day unregistered and sip show peers do >not show their IP and although these phones can make calls , they >cannot be called. They Sip show peers

Just got my new Budgettone phone, and I've got a couple of issues. Most important, it doesn't seem to be querying for the time via NTP. I put a sniffer on the line, and once it boots up the only outbound traffic it generates is an attempt to contact a TFTP server, which is programmed in as 192.168.0.168. . . Must it first find a config file (it's asking for "cfg.txt")

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Hi Just recieved the above phone Does anyone have sip.conf and extension.conf example for the SIP phone working with the FXS w100p and the FXO tdm400d any help would be appreciated Thanks Robb

Hi- I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to make sure it has dhcp enabled and it is. The display says no IP. I bought the phone but do not have any documentation other than the Wiki, but I am still at a loss. What could be preventing the phone from picking up an IP address? Any

Now I have searched around and not seen anything to do this. I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote

Hello all. I don't know how I got onto this list, but now I'm here I'd like to ask a question: Is there currently a way of implementing NUT with an Ovislink Chrome 1500 UPS? Is this even the right place to ask such a question? Thanks in advance and apologies if I'm in the wrong place. Cheers. Paul. -- Work: https://paulc.es

Here is a quick tally of the various things people asked for.. I'm going to go thru the list and weight the results based on my scale of 1-10. This is just a count of each "item", otherwords how many times that item came up. Some things I considered as bugs and lumped them as bug-fixes For the various requests for codec's I broke out which ones people where asking for.

Hello. I had grandstream working fine to FWD through my firewall. Now I want it to talk to the asterisk server. Did lots of reading, attempts but I keep getting registration errors even though I can call to/from the sip phone from an analog phone on a tdm400 card. Basically. grandstream = 192.168.1.70 asterisk = 192.168.1.1 The error I see is ;- -- Executing Dial("Zap/2-1",

hi .. i've just converted myself back to a newbie by trying to experiment with some new stuff .. I have connected two grandstream Budgettone 100 phones to my asterisk, and trying to experiment with them .. I am trying to get into the asterisk sample basically .. when I dial 1000 asterisk receives the call, but I do not hear any sound on the phone. Dialling from phone to phone direct (via

Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or

On 10/07/15 04:26, Charles Lepple wrote: > On Jul 9, 2015, at 9:43 AM, paul at paulcarmichael.org wrote: > >> Is there currently a way of implementing NUT with an Ovislink Chrome 1500 UPS? > > Not sure. There was a thread in February talking about an Ovislink Chrome 1000, but I don't think we ever resolved what was causing the "Device busy" error: > >

On 10/07/15 04:26, Charles Lepple wrote: > On Jul 9, 2015, at 9:43 AM, paul at paulcarmichael.org wrote: > >> Is there currently a way of implementing NUT with an Ovislink Chrome 1500 UPS? > > Not sure. There was a thread in February talking about an Ovislink Chrome 1000, but I don't think we ever resolved what was causing the "Device busy" error: Is there an