similar to: threeway calling

I'm running Asterisk (stable branch downloaded 2-Sep-05) on RedHat 9 and I have a TDM22B installed (TDM400 w/ 2 FXO and 2 FXS ports). Everything seems to work except threeway calling. I can establish a threeway call, but it uses up BOTH FXO lines. Note that I DO have threeway calling active with my Bell service. Here's a typical scenario: 1) Call 765-1574, 2) When they answer, press

How much can be the load (How much register and calls Asterisk can Handle simultaneously by asterisk) and what will be the performance of Asterisk (Call Quality) if all the users are on SIP only and uses same Codec, I have all three codecs loaded G.711, G.723, G.729) without media support i.e. ("canreinvite=yes"), Thanks & Regards Ritesh Jalan -------------- next part

I was just wondering if there was a way that you could have two calls on one line and switch between the two without initiating a threeway conversation? I would imagine that Flash is the way to do this, but when I Flash twice, a 3-way call is initiated. If I turn threeway off, then I can't transfer. Also, is it possible to hang up one of the calls, and then continue talking to the second

I think I found another interesting 'feature' with threeway calling. If you hang up while on a 3 way call with both parties on outside lines, Asterisk ends up removing the conference initiator and leaving the outside trunks bridged together. Is this a good idea? This could cause congestion problems on small configurations with limited outgoing lines. Maybe we should add an option to

I tried to add the databases from the localhost to the database server and changed the every /etc/asterisk/*.conf from host=localhost to host=192.168.10.10 (my dababase server) When I restart asterisk, I do not get any errors, but after a phone call I see: Jun 7 18:11:56 ERROR[7877]: cdr_addon_mysql.c:400 my_load_module: Failed to connect to mysql database cdr on 192.168.10.10 Or if I try

Hi All We are unable to hear any voice where as in tcpdum it shows that RTP is flowing both ways ERROR CONDITION --------------- -- Executing Dial("SIP/2001-f6c4", "SIP/2000|20") in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

Hi guys, I would llike to ask for your help on a problem I'm having with the cdr functionality. I installed asterisk 1.0.4, and asterisk-addons-1.0.4 and followed the procedures for installation and mysql configuration. Everything seems fina. The cdr_mysql module is loaded, and I get no error messages. But the cdr's are not going to the mysql database, they are going to a csv file in

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 setup a Meetme conference room and i notice the following behavior: if A calls confroom over PSTN channel 1 B call confroom over PSTN channel 2 C calls confroom over SIP/Ethernet then i have all of them talking and the media stream mixed by asterisk. However, if i hang up A, channel 1 is still ocuppied (i try dialing inbound again on channel and it continues to give a busy siganl) any

hi; does any body know what are the physical dimension of a digium care 400pm for example? thanks m.smadi

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

I have difficulties in interpreting high order interaction terms in high-way ANOVA. According to Introductory Statistics with R by Peter Dalgaard (Section 12.5), "The exact definition of the interaction terms and the interpretation of their associated regression coefficients can be elusive. Some peculiar things happen if an interaction term is present but one or more of the main effects are

What are the meaning of these Zap show channel output? Caller ID string: Owner: <None> Real: <None> Callwait: <None> Threeway: <None> Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax handled: no Pulse phone: no Echo Cancellation: 0 taps unless TDM bridged, currently OFF Actual Confinfo: Num/0,

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hi, We are using threewaycalling & flash transfers over a CAC channelbank. The following happens: Call comes in to my extension I talk to a party and press flash party goes on hold, I get get dail tone I dial internal number internal party answers I press flash once more we are now in a three party conference Or I hang up, and thus transfer the call. Thats fine, but.... What if the

I new in asterisk world so, please, forgive me if I say something stupid. At least, and after a lot of tryes, the isdn card seems to be registered: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Automatically generated pseudo

hi all, I am starting to learn ruby and using ruby on rails. I have one question. I want to create a webpage that has two sections. The right section has all the tags (stored in a MySQL table) and in the section I display websites related to selected tag. I am not sure how to do this as I have two different actions to be fired in every webpage. Any suggestion is most welcomed Regards,