similar to: Asterisk SIP + Grandstream 100 phone

hi, can someone who has used Budgettone phones tell me how to do the following: an incoming call comes in and is answered by the receptionist. she need to put the call on hold, speak to whoever the call is for, and either (after that) pass on the call, otherwise speak again to whoever was on the call and hang up .. so far i've got as far as a blind transfer by pressing transfer button and

Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi- I am attempting to setup my Budgettone phone for use with my * server and am having problems obtaining an IP address. I have checked the phones settings to make sure it has dhcp enabled and it is. The display says no IP. I bought the phone but do not have any documentation other than the Wiki, but I am still at a loss. What could be preventing the phone from picking up an IP address? Any

Dear all, I have a problem with constructing a nested loop. I have two matrices: pnr: 800 rows 14 columns where rows are 40x20 meaning that 40 rows belong to one of twenty objects in the matrix pnr mvp: 20 rows and 14 columns I want to: calculate a distance value with the first 40 rows of pnr but 20times with the 1st row of mvp then the next 40 rows of pnr and 20times with the second row

Hi, I installed Asterisk an all works fine exept for Grandstream. When I call with a softphone (ex X-ten) to a Grandstream (BudgetTone-100), I can make a conversation. = ok When I call to a softphone with a Grandstream I can pich up the call with the softphone but the Grandstream keeps ringing like on the other site you didn't pick up the phone.(even if you do so) It's the same when I

I upgraded from RC2 last night, but have a major call quality issue. Heres our setup: 1 FXS and 1 FXO card. Incoming/Outgoing calls via IAX trunking from our provider. G729 running between us and the VoIP provider. Two handsets, one BudgetTone 102 and a Cisco 7940G running the 7.2 SIP firmware. Both these phones are using ULAW to the server, and we have plenty of G729 licenses on the server.

hi there .. I asked sometime ago regarding getting a Budgettone working with Asterisk over G729. My system is quite simple, Asterisk server with 1 G 729 license installed, and 10 Grandstream phones. Only one of them needs G729, because it's on a remote link via an ADSL bridge. The rest run happily on G711 on a local network. I added the lines disallow=all allow=g729 to the sip.conf entry

In zfs_mount() function, when we process a mount by a regular user through the delegated administration, the comment states: /* * Make sure user is the owner of the mount point * or has sufficient privileges. */ This makes sense, but the code doesn''t match the comment. The code ensures that user is the owner of the mount point _and_ can write to the directory. Or does "has

--- "Chandra" <chandra@digital.com.np> wrote: >i have been using grandstream budgettone IP phones and they work fine >except that these phones times out after some hours.. i ahve seen that >the phones working ok are next day unregistered and sip show peers do >not show their IP and although these phones can make calls , they >cannot be called. They Sip show peers

I have an asterisk installation at a client, it's quite simple. Basically it's an asterisk downloaded from CVS about a week ago, with 3 Zaptel FXO cards (the digium ones) and 10 Grandstream Budgettone SIP phones ... Every now and then, especially when a call is ringing and not picked up immediately, Asterisk quits with a segmentation fault error. IT seems quite inexplicable, my dialplan

Just got my new Budgettone phone, and I've got a couple of issues. Most important, it doesn't seem to be querying for the time via NTP. I put a sniffer on the line, and once it boots up the only outbound traffic it generates is an attempt to contact a TFTP server, which is programmed in as 192.168.0.168. . . Must it first find a config file (it's asking for "cfg.txt")

Here is a quick tally of the various things people asked for.. I'm going to go thru the list and weight the results based on my scale of 1-10. This is just a count of each "item", otherwords how many times that item came up. Some things I considered as bugs and lumped them as bug-fixes For the various requests for codec's I broke out which ones people where asking for.

System info: Red Hat Enterprise Linux Server release 5 (Tikanga) Kerlen 2.6.18-8.el5 SMP x86_64 Samba version 3.0.23c-2 Eth0 && Eht1 bonded to bond0, 2Gbps. /etc/samba/smb.conf attached below... I?m seeing very slow transfers from Samba.... I?m not sure how else to describe it. If I try and copy a 4GB DVD image from the server to any Windows box (XP, 2003, 2008,

Ok, we got it working with 176x144 resolution. The following string produced an image that was not falling behind too much for too long - basically a little when it was getting TONS of movement, and then catching up fast. "65 58 110 0 1 1 90 8 120 1 1 " Now, we need to get this working on a 320x240 resolution, too (as sadly VP3 does not support 160x120. Seems to be a little senseless,

Hello, I have read the information on echo and SIP in the FAQ and I have scoured the mailing list for possible solutions, but as yet I have not been able to get rid of this echo. I have a Sip phone, Digium 4 Port FXS, and 3 Digium FXO cards installed into an asterisk server. If I call between the Sip Phone (Budgettone-100) and the 4 FXS ports everything sounds great. If I call out to the PSTN

Hi, I have used the Budgetone 102's extensively on Asterisk and found then quite reliable as long as you update the firmware. The GXP2000 is quite a mess at the moment as the current firmware does not support 3 quarters of the advertised functions and codec support is extremely limited. I have tested the unreleased latest firmware update for the GXP2000 and it's an incredible difference

Like most others on this list I also have some really annoying echo whenever a call goes out to the PSTN from a SIP phone... SNOM/Budgettone -> Asterisk -> X100P -> PSTN I have tried every echo canceler in the makefile and turned on and off aggressive suppressor etc. etc. etc. tried 32,16,128, and 256 bridgetaps and I can get it reduced to only a few seconds on the intro of the call and

Hello all, When attempting a classification tree using mvpart, I get the following error: > thesis2.mvp=mvpart(bat_sp~., data=alltrees.df) Error in all(keep) : unused argument(s) (c(TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE, TRUE,

Hello, We put the DirectShow filters on hold till the release of our software (next week) and will integrate it into the first patch. Till then, our customers will ahve to live with a VP31 vfw hack. Now, my problem is that I need to integrate the vp31vfw.dll setup into a MSI file, because spawning another installer is not exactly an option. Can anyone post a list of complete registry keys that