similar to: Grandstream and CallerID not working

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

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

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of

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

List Mom/Dad, could you please find in the list of subscribers the person 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:

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 --

Hello, I am very new to R and Time Series. I need some help including R codes about the following issues. I' ll really appreciate any number of answers... # I have a time series data composed of 24 values: myinput = c(n1,n2...,n24); # In order to make a forecasting a, I use the following codes result1 = arima(ts(myinput),order = c(p,d,q),seasonal = list(order=c(P,D,Q))) result2 =

Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems to be having problems. The Grandstream and sip.conf were set to RFC2833 now with that setting I get extra digits during "Mailbox" and "Password" phases. 222001 instead of 2201 for example. When both are changed to "SIP info" there is no problem. But what is the new setting "DTMF Payload

any messages in the windows 10 event logs, that could give some extra insight. according to https://social.technet.microsoft.com/Forums/en-US/7f5207cc-b202-47fc-bbb8-9ebe46a31961/network-logon-script-failure?forum=WinPreview2014General >\\foo.lan\netlogon should work. but, https://adsecurity.org/?p=1405 has some good info about the latest patch about hardening GPO. (which imo wil be

Symptom: Phone after about 15mins will stop functioning Problem: DHCP lease renewed but default route dropped Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is released It turn's out that these phones have a few issue in 1.0.3.81 firmware. The phone may stop transmitting packets if configured with DHCP, if DHCP is being provided by certain devices. Netopia

I inadvertently unplugged my Budgetone phone tonight and I think it went out and upgraded its firmware. It is now at 1.0.3.81. Does anyone know how new this might be? Suddenly things that have been working wonderfully before don't work--basically it seems to do the SIP stuff just fine but then all the RTP stuff breaks, even something as simple as retrieving my voicemail from a server on