similar to: * reports the incoming caller id but not the BT100

I've looked through the archives - and see questions similar to mine, but no answers. What, if anything, can be done to get the incoming Caller ID to be presented on the Budgetone's Caller ID display? In all other respects the phone+Asterisk seem to be extremely happy with each other.

No idea for this problem? Alex -----Mensaje reenviado----- From: Ing CIP Alejandro Celi Mari?tegui <alex@linux.org.pe> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] [Fwd: newbie DNS problem with BT100] Date: Tue, 22 Mar 2005 19:42:30 -0500 (Sorry, but my english is very bad) Hi I'm newbie with

(Sorry, but my english is very bad) Hi I'm newbie with Asterisk, but i was able to install and configure Asterisk with 3 budgetone 102 and 2 Handytone 206 and works fine for me. I have a problem and i don't see answer in forums: DNS resolution: First Day: ========== In configuration menu of the BT100 I use: DHCP SIP server: central.mydomain.com or 192.168.100.180 Use DNS SRV: Yes NTP

Hello, Has anyone gotten a Grandstream BT100 to work with Level 3's 3Tone? I've been able to get my extension to interface with it, but there is no sound and the call on the GS side terminates prematurely. Here is the relavent portion of the SIP.CONF [4007] ; Budgetone BT100 type=friend insecure=yes context=test-budget username=4007 fromuser=4007 callerid=4007 host=dynamic nat=yes

Hi List! I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems with * modules refusing to build I replaced the RHEL kernel with stock 2.6.10. Asterisk seems to be working but when I dial voicemail I hear nothing. When I hangup I see a message on the console that the calller did not specify a mailbox number so I guess voicemail app is working. The phone(Grandstream BT100) is

I did some searching in the archive, but found only one message with this same question and no answer. Hopefully it's a simple config problem. When the Caller-ID is delivered, it is surrounded by double-quotes, like this: "ATA-57 1" On long caller-id strings, the last character is cut off to make room for the leading double-quote: "BudgeTone 1234 instead of BudgeTone

I tried to get callerid working the normal way but the cid is never passed to the phone. It doesn't work untill I set SetCIDNum(0${PRI_NETWORK_CID}) in extensions.conf which I found in the wiki: http://www.voip-info.org/tiki-print.php?page=Asterisk+zaphfc Is this intended behaviour, or still a bug? It does work but it only shows one zero even though I have nationalprefix = 0

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration ====================== Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

Caller id is not in the configs that are supplied with zaphfc from bristuff, and it's not mentioned in the wiki. Is it possible to use callerid with zaphfc/bristuff? Thanks!!

Hi, I recently encountered a weired situation where my budgetone stopped working. My network is like this: Asterisk on Public IP ----------- ADSL NAT Router ----- GS01, GS02, GS03 on Internal IP We have an Asterisk server running with a public IP address, which serves as the master PBX. On a remote site, we have 3 Budgetones all having internal IP addresses assigned by the ADSL NAT router. The

is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami

Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone

I have a couple of Budgetones that I am playing with trying to get them to work with * from a remote network over the Internet (yes NAT joy!). My * server is in my DMZ and I have 5060 and my RTP range forwarded (UDP) to my public address (through a Cisco PIX). Internally, I can setup my budgetone, it registers and works great. I then have a Linksys router connected to another Internet

We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with

Hi list! I have some strange problems with Asterisk 1.0.5-BRIstuffed-0.2.0-RC5. Very regularly asterisk seems to lose connectivity with the ISDN line. If you try to call in you get the information tone that the number is not in use. Outbound calls do stil work however. Unloading the modules and reloading them and start/stop asterisk will solve the problem. Another problem that occurs

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks!

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco