similar to: how to config call waiting and three way calling

Do you have threewaycalling and transfer set in your zapata.conf? Here's mine (four TDM400's, seems to be working so far). I didn't do anything in my extensions.conf for any of these features (what confused me at first is the t and T options of the Dial application in extensions.conf are for transfers via the # key), when you flash you get another dialtone that works just like the

HI; I have an Asterisk with 10 "SIP" ip-phones, our pbx features are now: Voicemail and Call Transfer. How can I serve both "Call Waiting / 3 way calling" for our SIP Phones.?/ Appreciate Any Help Mohammad -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two internal analogue extensions off a TDM400P. If the first is busy, I'd like to ring the second. So: [incoming] exten =>s,1,Answer() exten =>s,n,Dial(${mainline},60) exten =>s,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Dial(${secondline},30)) But it doesn't work because * first tries Call Waiting on the main line. Here I dial out: -- Starting

Does anyone have an example extensions.conf section for initiating a three-way call? I don't see documentation on the syntax anywhere, and haven't been able to figure it out from the source. Thanks! Joel

I use cisco ATA 186 ( Version: v2.16 ) with sip protocol and asterisk as pbx. I need feature called as 'three way calling' or 'transfer with consultation'. Registering,calling and 'blind transfer' work fine. Is this feature provided by sip clients or by asterisk itself ? What I have to configure in ATA and what keys I have to press on my phones ? Three way calling is

I was wondering if anyone is able to use the three way calling features from their telco on the incoming FXO lines to transfer a caller back out to say a cell phone. I am currently moving from a Talkswitch to the Asterisk PBX and one nice feature they have is after 4 rings or so I can have the call transferred to my cell phone using the same line it came in on with three way calling. Just

I guess what I meant to ask was for a way to do it from within extensions.conf. Using either the Dial command or if there is another method to do the three way calling. >Press flash on your phone (asterisk will intercept that) and then when you >have a dialtone press *0 then asterisk will send the flash to PSTN line. > >regards >Martin > >On Thu, 28 Aug 2003, Carlton J.

I'm having a problem getting 3 way calling to work correctly using two outside lines and one extension. The two outside lines are connected to the X100P's and a standard model 2500 phone is connected to the TDM10. When I dial the first outside destination 9xxxxxxx, the call completes correctly. When I flash the hook switch and dial the second location 9yyyyyyy. The call doesn't

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be

I have a SIP phone that doesn't support three-way calling. Is there a way to do three-way calling from a SIP phone server side instead? TKS __________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

Hi guys, Does anyone know of a way where I can bring a third person in on my conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM and am speaking to someone now I want to quickly bring another SIP or IAX extension into this call so the three of us can speak to each other. I know I could do this by transfering the first person into a meetme then calling the second

Hi, this was my fault, you are right, i tried with a X-lite Professional and the conference (3-way call) is working now, i guess the phone BT-100 doesnt support it, i dont have a BT102D, so i can tell if it works too, bye -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of mmiranda@americatel.com.sv Sent: Friday,

Hi, i cant make a three way call using grandstream phones (BT-100) and asterisk using sip, is this supported or i need a zap interface? thanks

Hi, I would like to understand how those features (subject) work on fxs ports. Unfortunately I don't have a digium card with this kind of port, then any help will be appreciated. I tried to gather some information from google and this list history, but I still need some help. 3-way-call - As I could understand, when you are talking to A-person, you can press *flash*, call to B-person and

We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear clearly. Nothing in NY. Here's the sydney server: -- Accepting AUTHENTICATED call from <zoiperipaddr>: > requested format = speex, > requested prefs = (), > actual format = ulaw, > host prefs = (silk16|ulaw|gsm|g722),

if you have anyone questions about your service you can contact us at the support 978-418-7300 James Jones Broadvoice Technical Support ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Ben Wern Sent: Sat 8/28/2004 4:34 PM To: Asterisk Users Subject: [Asterisk-Users] Broadvoice BYOD Plans - 3-way and Call Waiting Can anyone who is using Asterisk

I've configured asterisk to connect a MySQL database for CDR, Voicemail and SIP/IAX2 peers. - CDR are reccorded - Voicemail config is readen directly in the database but SIP/IAX2 peers still have to be declared in sip/iax2.conf to make calls... However, when I restart Asterisk: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found ==

Hi all, I'm doing a 3-way ANOVA like this: summary(aov(formula('FP ~ (lum * obj * man)3 - Error(vp/(lum * obj * man)3)'),data=dataf)) But in the output I only get 1- and 2-way effects, like this one: Error: vp:obj:man Df Sum Sq Mean Sq F value Pr(>F) obj:man 1 1.5291e-34 1.5291e-34 5.7011 0.0542 . Residuals 6 1.6093e-34 2.6822e-35 --- Signif. codes: 0 '***' 0.001

I have 2 Asterisk servers connected via IAX2 connections. PBX1 is on the internet with a public IP Address - with PRI PBX 2 is behind a NAT router with IAX2 Ports forwarded 1-way audio is an issue with incoming and outgoing calls using the PRI. However whenever 1-way audio occurs, PBX2 can call PBX1 extensions and there are no issues. As well as a restart of asterisk on PBX2