search for: 3way

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

I'm having problems with conference calls (3-way) when I have my codec forced to g729 in sip.conf. I'm using Grandstream 2000s. If enable both g711 and g729 then 3 way calling and transfers work. I'm not sure why this would matter? Here's the error: Oct 13 13:54:45 NOTICE[31184] chan_sip.c: No compatible codecs! Any help is greatly appreciated!

...column per row is itself a data frame, and columns 2 to 4 will keep numeric values. The data frame contained in the 1st column will have 54 rows (with special names) and 4 colums (1st col is a response, cols 2- 4 are factors). Each of these data frames with the response/factors will be fed into an 3way linear model for anova. The other colums of the 1st data will hold the p-values. Basically running 7,000 anovas is very quick but the reformating of the data so that it is suitable for the anova takes a long time (45 minutes). So I'd just like to keep the generated data structure as a persiste...

...numbers without tieing up my lines. The following has worked successfully for me and I just thought I'd post it so if someone was looking to do the same they could quite easily. The feature you need installed on your lines is called conference-drop-transfer or here in canada it's know as 3way-call-transfer. exten => 533,2,Senddtmf,18665582273 exten => 533,3,Wait(2) exten => 533,4,Flash exten => 533,5,Hangup call comes in, caller dials 533 for external support line, asterisk Flashes, sends dtmf to dial out via the incoming channel, waits(2) seconds, Flashes back, hangs up,...

Hi, I have a TDM400P (31B) in a PIV 2.8, 512Mb ram, CentOS 4.3, zaptel 1.2.5 and Asterisk 1.2.7.1 and a couple of standard analog phones with a flash button. A hook flash works fine for setting up a 3way call. But pressing the flash button doesn't do anything. The zapata config is below. Anyone have an idea what I'm doing wrong? [channels] context=local usercallerid=yes hidecallerid=no immediate=no transfer=yes threewaycalling=yes canpark=yes echocancel=yes busydetect=yes signalling=fxo_k...

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

...tions about codecs: What codec does the Zap channel use by default? Can this default be changed, and to what? (g729 too?) What codec does meetme use? (I think this is ulaw, but asking to be sure) Can you use another codec, or does everything have to be transcoded to ulaw? Finally ... if I have a 3way call going, between 1 g729 caller and two other callers, do I need one or two available licences? (I'm guessing that zap doesn't do g729, and am wondering if I have an FXO caller and a local FXS person talking to a VoIP caller using g729, how it would work) Trying to nail down how all of t...

...-am.txt | 5 ++++- git-am.sh | 4 +++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/Documentation/git-am.txt b/Documentation/git-am.txt index 51297d0..4c65dba 100644 --- a/Documentation/git-am.txt +++ b/Documentation/git-am.txt @@ -13,7 +13,8 @@ SYNOPSIS [--3way] [--interactive] [--committer-date-is-author-date] [--ignore-date] [--ignore-space-change | --ignore-whitespace] [--whitespace=<option>] [-C<n>] [-p<n>] [--directory=<dir>] - [--reject] [-q | --quiet] [--scissors | --no-scissors] + [--exclude=PATH] [--include=PATH] [...

another issues with cisco 7975 I have phone registered on asterisk have 2 different issues on different versions of firmware, on 9-4-2-1S I have not working 3way conference, when I trying to connect second call, phone says ?unable to set up conference? and sending some cisco xml data to asterisk which cannot be handled, thats the problem, I know on firmware 8-5-4 3way conference works just fine 3cx phone system so must be same with asterisk, but with aste...

Good Morning, I would like to know if is possible to do a conference with 9 client with asterisk. The client is connecting to sever through lan, we think don't use PSTN or ISDN. Thanks, Alberto -- Alberto Carlana <alberto.carlana@virgilio.it> -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes

I have a x100p card that has call waiting on the line comming into it and then into *..... is there any way i can use call waiting on that line? Michael

...xpires: "3600" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # Enable VAD (0-disable (default), 1-enable) enable_vad: "0" # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (default) # Telnet Level (enable or disable the ab...

Other than cranking up logging to debug2, is there a way to better tune logging on a server to see if I am running into max sessions ? On FreeBSD RELENG11 I am periodically seeing connections being refused- 3way handshake not completing or completing and then FINs. Typically, I have a hundred or so connections at one time, but they can bounce up to a few hundred on occasion. Without leaving the server at debug2 logging level, is there a way to find out if I am hitting application limits vs OS limits ? A...

I didn''t think this was possible, but apparently it is. How does this work? How do you mirror data on a 3 disk set? This message posted from opensolaris.org

...blem is ? The 32-bit machine was just for a comparison, the plan is to run xen on that debian-amd64 machine. Btw: I did both tests on debian-etch aka ''testing'', using the package ''xen-linux-system-2.6.17-2-xen-686''. thank you VERY much! --- Rune Elvemo rune@3way.no _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

...e? 1. phone1 calls asterisk thru zap/fxo. asterisk gets callerid of phone1. 2. asterisk will callback phone1 using zap/fxo. 3. phone1 answers and is prompted for a number to call. 4. phone1 dials a number. 5. asterisks intiates a flash on zap/fxo. dials the number and then flash again zap/fxo for 3way call. i saw callback scripts but haven't found a sample script to do a flash and dial the number. i tried doing callback, flash, dial and flash but didn't work. tried callback, flash, senddtmf, and flash and didn't work either. any tips? thanks. __________________________________ Do...

...o 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 able to send DTMF to the authentication center. If done with the builtin * 3way calling the DTMF signals from the customer are not passed through the system. How do we have them signal the POTS line to use the 3-way feature on the line? We have tried the *0 on the dialpaid but it doesn't seem to work. Is there a setting somewhere to activate *0 or is there some other way t...

Has anyone been able to get 3way/Conference working with the snom200 and Asterisk. According to the documentation for the phones the option should come up when you have two lines active on the snom phone. Unfortunately, I don't see this option appear and I am now beginning to wonder if this is a limitation of Asterisk. Does an...

..., the # key (attended transfer) nothing happens. I can transfer the call if its coming in, but not if I made the call... the dial commands seem to be set, but the feature isn't working.. is there an obvious place to look to fix this one? I'm also wondering if there's an easy way to do a 3way call without using meetme.. any simple method like flash each caller to create a 3 way call? can't seem to get any of these to work at the moment. Thx as always Dan

...am.txt | 5 ++++- git-am.sh | 4 +++- 2 files changed, 7 insertions(+), 2 deletions(-) diff --git a/Documentation/git-am.txt b/Documentation/git-am.txt index 51297d0..4c65dba 100644 --- a/Documentation/git-am.txt +++ b/Documentation/git-am.txt @@ -13,7 +13,8 @@ SYNOPSIS [--3way] [--interactive] [--committer-date-is-author-date] [--ignore-date] [--ignore-space-change | --ignore-whitespace] [--whitespace=<option>] [-C<n>] [-p<n>] [--directory=<dir>] - [--reject] [-q | --quiet] [--scissors | --no-scissors] + [--exclude=PATH] [--include=PATH] [...