similar to: SIP phone choice?

Hello All. 1. I'm at first install ASTERISK (debian sid), and try to to make call by Kphone 3.1. For all my attempt I find in logs: Apr 18 16:05:17 NOTICE[81926]: File pbx.c, Line 999 (pbx_extension_helper): Cannot find extension context 'default' I check in extention.conf [default] section exist and has pattern which I send (for example 's', '1234'). In CLI I try to

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

I have the MusicOnHold feature working great when called from ATA-186 extensions. It's pretty cool. However, when I call from a BudgeTone-100 phone, no music is heard -- instead it continues the ringing feedback and acts like the call is unanswered. At the same time, I can call from (multiple) ATA-186 extensions and hear music as long as I like. How can I debug this? As far as I can tell,

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

Does anyone have X-Lite build 1059 working fully with Asterisk? The GSM Codec works very well now but we have problems when using G711 in that when I setup a ping between the two sites and then watch the latency, it steadily increases and starts at about 150ms and goes up to 2500ms within about 20 seconds. I have not investigated fully but I guess that its sending ever increasing size packets.

Just for the record and to possibly help with others who get BudgeTone phones. My asterisk box is behind NAT, and I use Vonage, NuFone, and iconnecthere for my "POTS backhaul." On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102. The BudgeTone definitely has issues wrt the RTP stream and NATting, although unfortunately I haven't yet been able to dig

I have a strange problem in that I have put a budgetone out on the internet that connects to my * server that's behind a firewall. They can call me I can call them and it works fine. However, I have setup a link to sipdiscount on my * server. If the budgetone user calls via my * box to sipdiscount all the budgetone user hears is silence and the called person hears silence as well when they

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

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

I'm sure this might not be the correct place to ask and I have done a Google but I can't seem to find anything that says there is a VSP that will work with * in the Ukraine. I have a friend that lives in Kiev and basically want a phone number there to be able to talk to him and have him call me. If anyone has any information on it and they are willing to share please advise.

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

Hello all, We've been using our Asterisk system live for about a month now and I'm looking to tuning a few things. First, is echo, I receive a fair amount of echo during the first 10-15 seconds of incoming calls. Next is a very weird problem. We have serveral Polycom IP300's and one Budgetone phone. It seems that if we unplug & move the Budgetone (which happens a fair amount

Hello All. I have few networks with Tivoli Provision Manager (http://www.ibm.com/developerworks/downloads/tiv/pmfos/?S_TACT=105AGX28&S_CMP=TRIALS ), Altiris (http://www.symantec.com/business/deployment-solution) and WDS (on W2008Stdx32). I successfully expand WDS functionally with pxelinux (4.03).I really like how pxelinux working. Server who respond on client request is on Windows 2008

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

Hello to all. I have a couple of budgetones connected to Asterisk server. I can establish calls using budgetone with no problem, but when I hang up a Budgetone, Asterisk does not detect the hangup. It seems that the communication goes on in spite of budgetone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend

Hello, everybody: My friends in Ukraine are starting a research lab at a national university and they asked what programs to use. I said "R" of course, and they then asked me 'what support does it have for Cyrillic'? i've done some snooping in the R website and all the references i find to foreign languages concern c and fortran, not Ukrainian or Russian. Since i'm an

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040709/2e9bd3f6/attachment.htm -------------- next part -------------- ? Hi there, This is shan. I need help regarding the grandstream Budgetone - 100 phone which i configured. My problem is: I can able to do call ---> Softphone(PC) ---> * ---> Grandstream Budgetone-100 but

Dear User, the message with following attributes has not been delivered, because contains an infected object. Sender = samba@samba.org (may be forged) Recipients = vladislav.smirnov@yabloko.ru Subject = Hurts Message-ID = [unknown-id] Antivirus filter report: --- Dr.Web report --- Following virus(es) has been found: infected with Win32.HLLM.Netsky.17920 Dr.Web detailed report:

The process for transfering a call with the Bugetone is as follows.. 1. Press "transfer", you will get a dial tone.. 2. Dial in the extension to wish to transfer to.. 3. Press the "Redial".. (on the newer phones this is the "send" button) You don't need the "t" option on your dial string to do transfers with a budgetone.. Later.. > With the right

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