search for: chagresventur

...n that is getting mail at uol.com.br or something similar and smack them for having broken Anti-Spam software ?? I'd be happy to see them just be removed until they fix it. cheers ----- Forwarded message from AntiSpam UOL <andersoncbr.sspam@uol.com.br> ----- Delivered-To: jmbrown@chagresventures.com Date: Wed, 24 Sep 2003 02:21:45 -0300 (BRT) From: AntiSpam UOL <andersoncbr.sspam@uol.com.br> To: jmbrown@chagresventures.com Subject: RE:[Asterisk-Users] initial review of Grandstream HT-286 ATA device X-UOL-Srv: T X-Spam-Status: No, hits=1.1 required=5.0 tests=CLICK_BELOW,HTML_50_60,...

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 ??

As I've said several times on this list[insert usual apology here], I still haven't received the last 20 of 100 phones I ordered over 2 months ago. If you get a hold of them please let me know MATT--- -----Original Message----- From: mikeu [mailto:mikeu@execpc.com] Sent: Saturday, January 10, 2004 12:41 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Chagres

Hi List, Could some kind soul post me a quick config that makes use of a Wildcat X100P when I do a show channels nothing is there lsmod shows the wcfxo and related drivers loaded and with no errors zap show channels is blank as well mucho thanks

Grandstream and Global IP Sound have inked a deal in which Global IP Sound will provide its royalty free iLBC codec to Grandstream. GS will integrate this codec into the BT and HT product lines

anyone around here have the ability to terminate a .NL phone number to IAX or SIP ?? if so please contact me off list. thank you

Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

Hi list, been playing with * for a bit now and have found at least two things that make it difficult. 1. priority numbers in extensions.conf This line numbering gig can be a PIA. While I under- stand there needs to be a way to branch control (ergo goto), it would seem that line numbering is tied to 1,2,3,4 as in exten => s,1,doblah exten => s,2,dofi exten => s,3 gogetalife exten

So I'm not getting caller ID via the X100P card. I've confirmed the PSTN is sending me caller id by plugging in a little third party box. Ideas, tips, greatly appreciated

Hi List, After reading a bunch of the docs, list post archives, it still seems that a clear definition of how to clock the T100P card is muddy. zttool says that the link is "INTERNALLY CLOCKED", does this mean the T100P is providing clock, or does this mean the T100P is getting clock from the T1 line side (ergo getting clock from the Telco) ?? If you have sync = 0 then zttool

Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please

Hi list, Well I really didn't want to see things get to this point, but Sherman at Sipura along with their President Jan F. leave me no other choice. SIPURA has been provided a letter from our attorney for Breach of Contract and damages. They have yet to respond. A quick background. 1. Sherman (SIPURA's Director of Marketing), stated that we would do a join press release for the Oct

Hi List, I had a wonderful meeting with GS's President last week and he is very interested in feedback on what top features, functions, bugs the community would like to see in upcoming firmware. Please keep in mind that adding new features take time to develop, test and such. So please rate your ideas on a scale of 1-10 1 = Nice to have some day 10 = Got to have it right now Things

Some guy on eBay is trying to sell the Grandstream Budgetone Phone 101 as the 102D. And to make matters worse he starts the bid at $90.00 Beware. http://search.ebay.com/search/search.dll?query=sip+phone&ht=1&sosortproperty=1&from=R10&BasicSearch= -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

I've only gotten to hear Music On Hold once. I am running the right version of mpg123 .059r as downloaded from mpg123.de and compiled locally So I'm looking for any help on getting this up and running. I can see on the console that the SIP phone is placing the call on hold, but there is no audio coming out on the other phone. Mucho, thanks john

Hi list, I've got a testbed running with the following config: 1. RH 7.3 linux machine 2. 2 Grandstream phones 3. 2 XTen soft clients When I dial voice mail I have a problem. here is the flow. 1. dial 8500 (the exten for voice mail) 2. enter 2600 via touch pad on the grandstreem 3. entire passwd via touch pad I always get an invalid login When looking at the Ast console I see that the

so I just built a new system, Slackware 9 based, as RH was giving me a major headache. after getting latest (today) CVS I'm noticing that the dial command is missing via the CLI. any ideas ??

Remembering that at one point I was a hardware geek, th GS seems to use the TMS320VC5402 PGE100 chip. According to TI's web site http://focus.ti.com/docs/tool/list.jthml?familyId=324&toolTypeId=32 there are a number of other Codec's available, including GSM ;) This information was figured out via already public sources of info. Now if they would use the DSP+RISC then you could

is there a clean way to have MOH (Music on Hold) source its audio from say a MS-ASF streem ??? got a radio station that wants to have their MOH come from their ASF based netbroadcast