similar to: Voicemail2 and RFC2833 DTMF

I have a problem with using voicemail on the Budgetone phones. When entering the mailbox and password, sometimes some keys will register multiple times (as shown on console when it says no such user in config file) and sometimes some keys won't even register at all. It seems totally random. Has anyone seen this problem? Any recommendations would be greatly appreciated. Thanks. Brian

Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my

Hi, When we use BudgeTone where the DTMF is set to "via RTP (RFC2833)" all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to "remote pick-up the call" through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Best

Hi, I've got a problem with some grandstram devices (namely a couple of budgetone 101 and an ata-486). The point is that, unless I use inband for DTMF, asterisk ignore the first digit dialed. Inband DTMF forces me to use A-law/Mu-law, which is not what I want. BTW, this appens after a Playtones(), waiting for user entering an extension. I've tried many solutions, played around with

Hi All I have 3 GrandStream BudgeTone-100's which connect to an * box with a HFC-S based ISDN card using ISDN4Linux. I have setup the BudgeTone-100's to use Early Dial which for calling between the three phones works well, but for the external calls using the following extension exten => _9.,1,Dial(Modem/g1:${EXTEN:1}) Only the first digit is dial on the ISDN Line. Does anyone know of

Hello, hope somebody can help. I have setup a queue which maps to some Budgetone SIP phones. When a call is answered, the # key to transfer a call does not work. Everything else regarding the queue works fine. Is there a way to activate it? Maybe something like the t option in the Dial application. Thanks in advance, Christian.

Hi, Just playing with a couple of Budgetone 102 phones and they are pretty good for the price. The only problem i'm having at the moment is when I get a voicemail on the Asterisk box the LCD flashes. Dialing *98 goes to the VoiceMail Manager, and asks for mailbox, I enter 201, then asks for password, enter my voicemail password set in the Extensions -> webadmin, then hit the

Hi, I've have followed through the help docs in trying to get an initial setup going with two phones and the asterisk server. Firstly, all I'm trying to do is get the two phones actually talking to one another VIA asterisk.. I've added this to sip.conf: [phone1] type=friend host=dynamic defaultip=192.168.1.106 ;username=blah ;secret=blah dtmfmode=rfc2833 ; Choices are inband,

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

Hello, I have a problem with asterisk and Grandstream BudgeTone-100. With default configuration everything works (in anonymous mode and fixed IP), but if Im trying to enable registering, it dos not work. I used 'sip debug' and verbose level 10, nothing happens if I switch telephone on (no messages about bad auth etc). As I understood, after switching phone on at first it will try to

A few weeks ago Brad posted his patches to the mailing list: http://www.universaltime.org/~brad/vmail/ But I can't find his email address... does anyone happen to have his address. I hope he would be willing to see if Mark wanted to add those options to the CVS. I think you need to fill out a disclaimer and post it to bugs.digium.com.... bkw

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Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk

We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone seen/dealt with the Grandstream BudgeTone-100 producing double dialing with asterisk? Thanks JDT Example: (password input = 101) -- Playing 'vm-password' -- Incorrect password '11001' for user '101' (context = <any>) -- Playing 'vm-incorrect' -- Playing 'vm-password' -- Incorrect password '110011' for user

We are experiencing intermittent DTMF problems here, with the following setup: ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF). I am running Ubuntu server 10.04, but Asterisk is compiled by us and not installed from the software repository. Essentially, DTMF works for some time, but at some point it simply stops and the point at which it stops appears to be random. Using RTP debug, I

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

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