similar to: Parking call problem

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

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 confused as anything by this bug. I'm hoping that it is just something screwy in my config. I have 1 Cisco 7960 and several Grandstream BT101 & 102's, and a Digium TDM31B. I'm running the latest CVS (CVS-HEAD-05/27/04-17:22:40) of both Asterisk and the Zaptel driver on Fedora Cora 1. When I make an outgoing call on the Cisco phone, everything works fine. I'm

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:

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

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

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

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

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, Is there a way to drop the ANSI colour strings from log files? In particular, I've got JSON logging throwing logs over to ES, but they have the ANSI colour escape sequences. Ideally I don't want to lose coloured logs from the console though, and I can't "see" a way to do this. Ast 16 at the moment… Andrew -------------- next part -------------- An HTML attachment was

Hi, I realise this is an old question, but I’m struggling to get my head around it. The ERD suggests that endpoints can link to multiple AoRs In what situation would you actually use this? Given that mapping of inbound calls is primary done to the endpoint, it looks to me like most of the scenarios where this might be beneficial are actually not possible? One example I had envisaged was being

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

Hi, This is not a big issue… I just noticed a build script that was pulling the 16.11.1 release from https://downloads.asterisk.org/pub/telephony/asterisk has started to fail and went to investigate (I need to test a patch for a bugifx). It looks like when the 16.12-rc1 RC was released the 16.11.1 was pulled from the current release directory. I realise it's probably better practice to pull

Hi, I'm trying to find (with some desperation now) a decent web based or application based UI that integrates with an Asterisk based PBX and is designed for a Serviced Office environment. Key features we're looking for: Concept of a client with multiple staff/associates Recognition of inbound DIDs with display of company details and greeting Support for multiple receptionists Some

I am getting this message: Failed to authenticate device <sip:2010 at X.X.X.X>;tag=149853321 for INVITE, code = -1 but it does not report the "connecting" address. Who is failing connecting ? I either need to block someone or fix something - I'm thinking block - but I dont know who. How do I found out the connecting IP? Jerry -------------- next part -------------- An HTML

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

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

Full_Name: Chris Yager Version: rm162 OS: Mac OS 9.2.2 Submission from: (NULL) (67.117.145.197) As a new user I was running A sample session from Appendix A of An Introduction to R. 'mm<-read.table("morley.tab")' failed. Note that the prior command, 'file.show("morley.tab")', succeeded. Problem does NOT occur in version rm161. NO other bug