similar to: Grandstream BudgeTone 100

How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]

I'm trying to use H323 for the first time so please forgive me if I've made a mistake here. I have downloaded and compiled the latest versions of pwlib, openh323 and asterisk. I have dtmfmode=inband in h323.conf, but the remote system is not hearing the DTMF. Running a trace reveals the following... 1:08.398 ThreadID=0x00022012 h323.cxx(4594) H323

Hi, My company wants to put a SIP address on their website. The idea is that potential customers can call that address and will be forwarded to our main switchboard. It's fairly easy in theory because my asterisk server has a real IP address, so any calls to sip:<number>@asterisk-server.mycompany.com should connect just fine (except currently it will be blocked by the firewall). Our

What's the secret to getting sound through Xlite? The SIP messages all look OK to me, but the sound isn't coming through. It was trying to use GSM, so I searched the archive and tried: disallow=gsm allow=ulaw Now it says that it's using ULAW but I still get no sound in either direction. Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.com>

There's something that I want to set up in our lab for testing purposes, but I'm not sure how to do it. I would like to be able to call an asterisk extension, and then enter a SIP address using DTMF, and then have asterisk make a SIP transfer to that address. For example: If I dial <extn> followed by 4444*192*168*0*10*5060 I would like to be transferred to

I updated to the latest CVS yesterday, from a version several months old. On one of my extensions, I have an AGI script in priority 1. Previously, the AGI script would run and when it terminated, asterisk would move on to priority 2 and connect the call. But now, when it terminates, it starts all over again in a continuous loop and never gets to priority 2. Do I need to update the priority in the

If I try to create a clearcase view on a Windows 2000 Server share then it creates a subdirectory tree. If I try it on a Samba share, it creates the first directory correctly with the right user mapping and permissions and then returns permission denied. I am able to manually create files and directories on the samba share using explorer, so I guess that clearcase must be using some different SMB

We have an h.323 based IVR platform. When we make a call to it using an h.323 phone, it can see the callers number (ANI), but when we make a call to it via asterisk, the call goes through OK, but we don't get the number. How can I make this work? h323.conf ======= [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=alaw dtmfmode=inband [ivr] type=h323 context=default extensions.conf

I am setting up a small Asterisk system for use at home. I have one Budgetone 101, one Cisco 7960 and two Xten lite softphones. So far everything is working expect for an issue with the Budgetone. When a call is placed between the Budgetone and any other phone, the call is setup and sounds good. If I hang-up on the Budgetone, everything is ok. If I hang up on the other phone, the Budgetone

Hello guys, I ask you to share your experience with your BudgeTone 100.... I have my asterisk @ work and I've bought a GrandStream BudgeTone (SIP phone) and I usually use X-Lite I have plugged my BudgeTone into my home network because I want to be called even at home. I succeed to register my X-Lite with Asterisk from home but I can't do that with my BudgeTone. (I don't know

Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ

Hello! I am considering mass deployment of Budgetones 102. According to their website, remote provisioning (configuration via TFTP) is possible. Anyone has experience with this? Is this really working?

Is anyone using this combination successfully? I have a dell 500sc running rh9 and asterisk 1.0rc1. It is configured with an x100p. I have a Sipura SPA-2000, laptop with Xlite and a Grandstream Budgetone BT-101. I have signed up with Voipjet (they use iax2). I also have an FWD number via iax2. I can sucessfully call back and forth to all devices via zap, sip, and fwd. I can successfully

Hi, I'm considering giving the Grandstream BudgeTone-102 phones a try. I've been using Cisco 7960's to date, but the low cost of the Grandstream phones are hard to ignore. I have two questions: 1) Does the message waiting indicator on the BudgeTone's work with Asterisk? 2) The one line 12-digit LDC concerns me a bit. Is the LCD able to display both the CID number and name on

Hi Folks, Since there isn't a grandstream forum AFAIK I guess someone here may be able to shed some light on this. Apologies if this is viewed as offtopic.. I think I may have killed the firmware on my Grandstream Budgetone 101. I found a source for the 1.0.5.30 firmware and made the files available over tftp. The phone downloaded the files but now doesn't boot and hangs with a blank

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

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

Maybe I'm loosing my mind but I've just noticed that if I put a call on speakerphone and I press speakerphone again it hangs up the call, you would expect it to take the call off speaker back on to the hand piece. I'm using V 1.0.5.22 firmware. Is there any other way to turn off speakerphone I'm missing? Cheers, Dean -------------- next part -------------- An

I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32 C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.20 25 Jan 18:25 Error Aborted 192.168.16.32 C:\Program Files\TFTP

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