similar to: PPP over T100P: Using a subset of channels does not always work correctly.

Hello! I am trying to set up multi-link PPP using two T100P cards in one machine, and 1 T405P card (the 4-port one) in another machine. I have previously been able to get PPP working between the two T100P cards in separate machines.... The 4-port card seems to be my problem currently. I am trying to use the tor2 driver (from a fresh CVS download this morning). When I load the driver (or run

Hello! Some time back, I added the Speex protocol to my version of VOCAL (www.vovida.org, VOIP tool). Recently, I also added PESQ (automated voice quality testing algorithm) to my tool and have been running some tests on a clean network. The source file is a woman reading some phrases meant to test various aspects of codecs... Speex has a respectable result of 3.67 Some other codecs I've

Francois Menard (Mailing List Account) wrote: > > did you try speex in wideband mode ... what bitrate? > > f. 15kbps mode. It does significantly better with a different speaker (the male does better than the female), as well. I'm considering purchasing a commercial PESQ license and wrap the PESQ software in a server application. I could then allow other folks to run PESQ

B. Mitchell Loebel wrote: > Hello LWATCDR: > > OK with me, but 1) I'm a Microsoft stockholder and 2) I'm not going to > stand still for people trashing Capitalist successes!! 'Nuff said. How about if I sell you a copy of speex. Buy low, sell high seems like a good capitalist plan, and then you could feel good about using Speex! Ben -- Ben Greear

Hello! My company makes testing software that, among other things, can be used to generate file-system traffic. In order to better support Samba, I need to make a few changes to the way samba mounts file systems: 1) I need to be able to specify the local IP address for the socket connection. This will allow me to bind different mounts to different local interfaces. This information will

Hello! I'm about to try to integrate SPEEX into the VOCAL project. If anyone has any pointers as to the best way to do this, please let me know. After reading the speex api man page, I have a few questions: 1) To encode, it appears I need an array of floats. If I am playing wav files, what is the best way to turn these into something speex can encode? 2) Are there any commercially

This portion of the RFC is gramatically incorrect and confusing: The RTP payload MUST be padded to provide an integer number of octets as the payload length. These padding bits MUST be all zero. This padding is only required for the last frame in the packet, and only to ensure the packet contents ends on an octet boundary. <p> -- Ben Greear <greearb@candelatech.com>

Here's a simple way to crash brctl (Fedora 8's brctl and tonight's git tree): [root at grok2 brctl]# ./brctl addbr bridge [root at grok2 brctl]# ./brctl show bridge name bridge id STP enabled interfaces . Segmentation fault (core dumped) Take it easy, Ben -- Ben Greear <greearb at candelatech.com> Candela Technologies Inc

Hello! I recently started testing curl's tftp options, and found that it crashed when trying to upload files using tftp because it didn't handle OACK properly. So, I'm attempting to fix it. Now, I have an endless loop: The client (my hacked curl) sends TFTP write request with options, server responds with OACK (tsize, blksize) client sends data packet, block 0 BUT: server responds

While working on adding support for local-binding, I found this piece of code in libmsrpc.c. Maybe I'm confused..but should that strcmp maybe be compared to != 0? /*change the server name in the server handle if necessary*/ if(srv && hnd->server && strcmp(hnd->server, srv) == 0) { SAFE_FREE(hnd->server); hnd->server = SMB_STRDUP(srv); }

I'm trying to compile speex on an x86_64 (Opteron) platform, but since I want this resulting code to run on regular 32-bit x86, then I am passing the -m32 flag to gcc. It so happens that the libtool application included with my version of speex (1.0.3) does not handle this correctly and fails to create libraries. I fixed this by changing the configure script to just use 'libtool' and

Hello! I'm working on adding VAD (silence suppression) to my branch of VOCAL. I've read through the Speex docs and see mentions of VAD/DTX, but I am not sure quite how to use it. For instance, on the transmitter side..do I need to detect the silence somehow and bypass speex encoding, or can speex tell me that the packet is all silent and should not be transmitted as part of it's

Jean-Marc Valin wrote: >>I'm about to try to integrate SPEEX into the VOCAL project. > > > Good. Just a detail, but the correct spelling in "Speex". > > >>1) To encode, it appears I need an array of floats. If >>I am playing wav files, what is the best way to turn these >>into something speex can encode? > > > Speex version

Jean-Marc Valin wrote: > AFAIK the padding changed with the latest draft and there's now a > function in speex_bits that does it all for you (and it's also called by > speex_bits_write). What that does is that it inserts a special > terminator, which allows a variable number of frames per packet without > any overhead. Does the stable release support the new draft? Is

(to mods: sorry for any extra work I've put you through. Posting from my work account, as proper) Hello all, I'm back. I've got that lucky feeling that this will all work out very soon, and I hope that that is true. The issue: In my * server I have a t100P, connected with a T1 crossover cable to an Adtran TA750 with 2 FXO cards. After I do 'modprobe wct1xxp;ztcfg' the alarm

I am having some difficulty getting a T100P card to work with my PRI. When I attempt to make an outbound call via: exten => 1004,1,Dial(Zap/g1/NPANXXXXXX) I see the following on the asterisk console: -- Executing Dial("SIP/sbruton-b8ce", "Zap/g1/NPANXXXXXX") in new stack Apr 21 08:18:48 NOTICE[16401]: app_dial.c:554 dial_exec: Unable to create channel of type

I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

Hi, I have two t100p cards installed. One card connected to the PSTN, the other card is hooked to the Definity PBX with a T1 cross over cable. The card connected to the PSTN works fine. The problem is with the card hooked up to the PBX. The light behind the card is green, but when I dial from this card I get an error. Its says it could start the D-Channel. Could someone help me out with this

I am having trouble getting my T100P and ADIT 600 talking. I have 2 FXO 8 port cards in the ADIT and a T1 controller card. I have a T1 x-over cable connecting two together. Here are copies of the relevant bits of my zaptel.conf and Zapata.conf Zaptel.conf span=1,0,0,esf,b8zs loadzone=us defaultzone=us Zapata.cfg context=default signalling=fxs_ks group=1 channel => 1-24 My