similar to: Grandstream BT102 Busy signal on hangup

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 : actual measured at 8.86 mbps. The

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 Bruce Komito High Sierra Networks, Inc.

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/attachment.htm

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: Stale nonce received

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)

Hello all, I am experimenting with Speex in a mobile VoIP application, and it seems that it is worth stuffing more than one codec frame into a single RTP packet; mainly, that sending several frames per packet relieves the underlying network socket connection, which is notoriously problematic in mobile devices. RFC 5574 defines the exact way how to put multiple Speex frames into a single RTP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On Tue, 12 Apr 2016, Bostjan Skufca wrote: > On 12 April 2016 at 10:23, A.L.E.C <alec at alec.pl> wrote: > >> I don't know dovecot's code, but I suppose it uses uidlist file to get >> mailbox statistics that it returns as EXISTS, RECENT, UNSEEN, UIDNEXT, >> UIDVALIDITY, etc, which are required by IMAP standard. I

Hi Marian, I took a brief look at RFC 5574 and as far as I understand, you can simply do it like this: SpeexBits b; speex_encode(state, some320bytePCM1, &b); speex_encode(state, some320bytePCM2, &b); After that, get the encoded data with speex_bits_write and put it in your packet. Mark -----Original Message----- From: speex-dev-bounces at xiph.org [mailto:speex-dev-bounces at

Last month I moved a 1.4.x Asterisk install to Asterisk 11.5.1. Everything is working well, until I noticed that Caller ID between facilities are showing properly, on the phone display, until the handset is picked up, then it's displaying NAME NOT FOUND. I do database lookup against extension number (Of remote PBX) and use that as the MySQL key to pull name. Snippet below: -- Executing

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

Jeff, RFC-5574 is standards-track: http://tools.ietf.org/html/rfc5574 so, while it's not an approved standard, it's more standardized than a lot of interoperable traffic on the internets these days. The RFC specifies packetization guidelines, which is basically that you put one or more frames in a packet, and then pad the rest with 0 bits until you have a while number of octets.

Jeff, It's in the manual: http://www.speex.org/docs/manual/speex-manual/node10.html (table 3 and 4). However, if you're asking this, you're probably trying to do something wrong, or the hard way. You probably shouldn't be taking speex output, and trying to "count bytes". If you are using the API, then you will just get the bits out, and then you'll know how

Hello, *Background:* The RFC 5574 suggests the RTP payload format for the speex codec. The payload formation is straight forward; the encoded frames are to be concatenated one after another. Once we have appended desired number of frames, we have to pad the stream with 01111 sort of sequence to ensure that payload ends on a octet boundary. *Observation:* I am using the speex encoder at 2150 Kbps

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

Hi, I swear I have read almost all the posted messages about this issue, but it's evident I couldn't find an answer (surely esay) to my problem. What I want is the following: Make 8 days aggregates from a daily series like this (dput output): structure(c(6.91777181625366, 0.79051125049591, 9.00625133514404, 9.86966037750244, 14.4326181411743, 3.70155477523804, 9.67768573760986,

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration

Hi, The Speex bit-packer already does everything you need. Just call speex_encode() multiple times to encode multiple frames and call speex_decode() multiple times to decode these frames. Jean-Marc Quoting Marian Kechlibar <marian.kechlibar at circletech.net>: > Hello all, > > I am experimenting with Speex in a mobile VoIP application, and it seems > that it is worth

Have you tried typing "speex rtp" into google code search? It gives lots of examples of real applications which do exactly that. http://www.google.com/codesearch?as_q=speex+rtp -SteveK On 11/1/10 1:13 PM, "Jeff Ramin" <jeff.ramin at singlewire.com> wrote: > >Thanks again Steve. I'll search for the term you mention below. > >What I really want is to

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