search for: bt102

FYI for anyone using the dual ethernet ports on a Grandstream BT102. I'm using a BT102 connected to an HP2524 10/100 switch, which has an asterisk box connected directly to it. No VLANs defined or in use. Measured bandwidth: PC -> HP Switch -> Asterisk : actual throughput measured at 94.1 mbps. PC -> BT102 -> HP Switch -> Asterisk : actua...

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA B...

Just got my (10) BT102 phones, flashed them to 1.0.5.20 and all work. No duds at all. Not a bad little phone at all. Doug

I have an * server running 1.0.9 on a FC3 machine. I connect around 44 BT102 phones to it and 6 Sipura 2000 units. Everything is working great but lately I have seen the following error: Aug 4 13:33:24 WARNING[31907]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:4000@148.235.174.85>' Aug 4 13:33:24 WARNING[31907]: chan_sip.c:4826 check_auth:...

Would anyone have a copy of the latest firmware release for the grandstream BT102 phone? seems grandstream no longer offers it on their website (of if I missed something a link would be much appreciated.) Thanks, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090504/9ba2730e/a...

Hey everybody, I've been playing around with Asterisk (Current CVS Stable dated: Asterisk CVS-v1-0-11/23/04). I've purchased 2 GS BT102 SIP phones. Both have been upgraded to firmware 1.0.5.18. I've also have installed on my desktop and laptop, X-Lite. I've been using these to learn how to setup Asterisk. I've got a Wildcat X100P on order and will be here next week. My problem seems to be with the BT102s. I can...

Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit over the top for simple users. What suggestion...

Hi I was wondering if there are any special settings that I need to be able to transfer calls. Whenever I press the 'recall' button, I just here a click, and no ring-tone to transfer. in my debug log I get this : -------------------------- Feb 24 09:09:27 DEBUG[19216]: Exception on 10, channel 1 Feb 24 09:09:27 DEBUG[19216]: Got event Pulse Start(14) on channel 1 (index 0) Feb 24

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

...call on their DDI), meaning that call in progress had to be terminated.&nbsp; Incoming call (i.e. call that was on call waiting) seemed to be fine, however.<br> <br> Running latest * and CAPI and using Eicon Server 4 port ISDN (BRI) card on RHES.&nbsp; Mixture of 5 x SIP phones (BT102) and 4 x analogue cordless phones (all connected to Digium TDM400P).<br> <br> Originally suspected problem was to do with BT102's (i.e. SIP handsets), but problem has since occurred with analogue phones as well.&nbsp; <br> <br> Appreciate any help with this, as sourc...

I have today reported a bug with the latest channel.c (1.134) that affects music-on-hold for the first user in a MeetMe room when calling from a Grandstream BT102. The music is broken up about 5-10 times a second. It doesn't happen when calling from Firefly. It is also fine on both clients with 1.133 of channel.c. I am using the ALAW codec. Mark at Digium can't reproduce the problem, although I don't know a) whether it is specific to GS phones,...

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

...nec electra elite 192 with a t1 card; and am looking for suggestions as to integrating them (can't throw out the system yet!). I have a fully working asterisk server -CVS-04/27/04-19:01:05- (found a hp d220 for $350.00!), 2 digium t100p cards, a plain t1 with loopstart signaling, and 2 working bt102 grandstream ip phones (thanks again Matt for your "start from scratch" article). This is what I'd like: t1 (loopstart)24 channels | | Asterisk t100p #1 | | Asterisk t100p #2 (best signaling option to NEC - e&m wink, ls, pri? it will do any of thes...

Hi, Has anyone experienced intermittent echo issues with voipjet lately?

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The ip600 is running v1.5.2.0054 and works very well in all other respects. Any thoughts on how to diagnose this? (Its definitely not...

...rying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with function keys is extension 690, the other extension (691) is just a BT102 so it doesn't have any function keys to program. When extension 691 is dialing out, or receives a call I want it to just tell the snom190 on ext 690 so the light shows up. (Soon as I got it going here I have a live system I will be setting it up on). Thank you to anyone in advance for the he...

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

Hi there, what phones are available that have two ethernet ports? I want to do some cabling at a new installation and i heard there are such phones (SIP i guess) out there. That way i dont have to run two cat5 to the user desktop. I think 3COM had one but can't find the web site reference for the two port phone thanks, erick

I've just noticed a strange behaviour with a MeetMe conference. I have a pair of phones (GS BT102) on my desk, and dialled both of them into a conference on speakerphone. If I spoke or made a sound, I heard it replayed from both speakers together a split second later, as expected. I went away for about 15 minutes, leaving the conference running. When I came back any sound I made came back out...

Hi, Really appreciate if someone who got astcc working lists the steps to make it work. I've got it installed and using the gui could get the database created. Would like to know how those two .conf files be populated and some pointers to the important fields in the database. Thanks Sathya -------------- next part -------------- An HTML attachment was scrubbed... URL: