similar to: Latest CVS, Grandstream and Zaptel bug?

As a follow up to my previous post, I have now identified what is causing the bug with the grandstream phones. When the line faxdetection=incoming is in the zapata.conf file, the grandstream phones will not ring, nor connect a call to the zaptel interface. Can anyone else confirm this bug? I'm going to play with the different options (incoming/outgoing/both) to see if it makes a

Hi, I'm not sure if I just missed something somewhere along the way, but I noticed while I was going through the CVS logs that there is an option in the wcfxs module to set an "opermode" - which apparently might help with echo issues around the globe (like the ones I'm seeing - some times). So I'd love to use it and give it a go. Apparently, there are some options that

Hi, I know we've talked about this phone to death. I have pretty good voice quality, with and without WEP enabled, using the G729a codec and DLink & Netgear access points. I am facing one obstacle that is driving me insane. Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold,

Hi, I don't know if this is possible - but can I set up asterisk to answer the FSO line after one or two rings rather than four? I haven't (yet) found a configuration variable to let me do this... Thanks in advance, Andrew _________________________ Andrew Yager Real World Technology Solutions Real People, Real SolUtions (tm) ph: (02) 9945 2567 fax: (02) 9945 2566 mob: 0405 15 2568

OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:

I been having a issue with call parking. I can park calls from internal extensions. But call from the outside can not be parked. When I recieve call from the outside I press the # key and nothing happens. Does any one have any thoughts? P.S. I am allowing the to be transferable. James --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com).

Just shoved my 7960 onto one of my clients networks. I'm talking from outside the office to them (they are using a GS BT101). I'm getting the following message repeatedly in the log, whenever someone talks. Jul 17 12:29:40 NOTICE[96402352]: frame.c:120 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Both phones are set to use the G729

Hi, I installed Asterisk with CAPI support. Everything works fine while starting Asterisk, but when a call comes in Asterisk hangsup the call after two times of ringing. The output is like: Jun 24 22:19:49 NOTICE[1082178480]: chan_capi.c:1931 capi_handle_msg: CONNECT_IND ID=002 #0x011d LEN=0048 Controller/PLCI/NCCI = 0x101 CIPValue = 0x10

Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. We need to set up our system to route incoming 56K data calls to the PPP daemon on our Linux box or to our

Hi, Just thought I'd let you guys know about my time in compiling on Panther. Not being the world's greatest Makefile expert, and after applying the patch suggested on this list (and modifying it so that the 7.2.0 was replaced by a 7.3.0 - for the latest Darwin kernel) I still had issues with my system requiring endian.h and byteswap.h . Being ignorant of most things, after much

Hi, I'm trying to get a fritz working in our asterisk box, however I'm getting the following error and was wondering if anyone could be of any help: -- Executing Macro("SIP/2002-5501", "enum-call|call|0405152568") in new stack -- Executing Goto("SIP/2002-5501", "call|0405152568|1") in new stack -- Goto (call,0405152568,1) ==

Hi Everyone, I'm one of those "newbie" users that really doesn't know what is going on. And in stark contrast to the posts you may read, I do actually think that when you post on this list, you get a reply. In fact, I know you do. I have had a quite nice run setting up asterisk, when I've had problems I've looked at the Wiki or Google, and then when I can't find

I'm trying to see if this is even possible. When you dial ext 2000 I want it to ring my sip phone for 20 sec then call my cell and let it ring for 10 sec if I do not pick up the call on my cell I would like it to go back to * and leave a voice message for me. Here is what I have so far in my extensions.conf Everything works except the call will not go back to * after the 10 sec rule has

When I dial a SIP phone which is specified in the sip.conf, but the phone is not connected, Asterisk gives the message "The user at Extension XXX is on the phone ...." Shouldn't the message be the unavailable message? Is there something wrong with my set up or is this a "bug" with Asterisk? Simon Brown

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

I've done all the googling I can on this, and have come to the conclusion that a Grandstream BT101 can be abused to be a door phone. Could someone with access to one, confirm that the following is possible? Researched: 1. When set to auto-answer, dialing the phone will result in a short beep and instant speaker-phone connection. 2. When pressing the "message" button while

I've seen lots of stuff go around about Grandstream firmware levels (in my case specifically the BT101/102). I'm just wondering what the currently accepted 'best' firmware version is to use? After seeing stuff going around about buggy firmware I want to know what I'm getting into before upping past my current 1.0.5.11. It's relatively stable, and the last thing I want

I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s) to go on permanent busy. Dialing that phone gives: -- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack

Hi All We have Grandstream 102's running ver X.18. When hanging up after a call has been made the grandstream seems not to disconnect the call and when you put the handset down the phone rings only to pick it up and be on the same call. This is happening quite often and gets very irritating. Can anyone help with this? Regards Doug

I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com