similar to: Conference Hangup

Displaying 20 results from an estimated 10000 matches similar to: "Conference Hangup"

2007 Jul 16
0
Dial and option G
Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1) [fromiax] exten => 9999,1,MeetMe(9999,qdxAa) exten => 9999,2,MeetMe(9999,qdx) exten =>
2007 Feb 09
0
Conference & Page question
Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option "d" (full-duplex), but I need to make ringing the phone in intercom. Now I set auto-answer=6 but after first ring the phone hangup the call. There is a way to using
2007 Oct 01
0
Park problem on IAX2 channel
Hi all, I have 2 asterisk box connected with IAX trunk. One box have connected a SIP phone and the second have a TDM card with one analog phone. When from SIP phone I try to park the call from analog phone with #700 the call is correctly parked but in the second asterisk I see this log: -- Executing Dial("Zap/2-1", "IAX2/CTM1/STI1|30|rjtT") -- Called CTM1/STI1 --
2007 Oct 05
0
Asterisk translator issue?
Hi all, I have a network with some asterisk in trunk with IAX2 and some SIP/ZAP phone connect to this *. In every call I need to use only alaw codec so in all conf file I have set disallow=all and allow=alaw. I try also to make some tuning of my environment removing unused codec and application. If I remove the codec_ulaw.so when I try to call I see this: [Oct 5 12:15:33] WARNING[16637]:
2007 Oct 29
1
Realtime & context
Hi all, I use asterisk with realtime features for extension, sip and iax. In extensions.conf I have put these lines: [from-internal] include => parkedcalls switch => Realtime/@ [fromiax] switch => Realtime/@ There is a way for put in my database the context also? Now if I want to add a new context I have to modify the extensions.conf with: [newcontext] switch => Realtime/@ but I
2006 Nov 22
1
qualify=yes
hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? I want to view whitch voip-phone is connected but I don't want to DOS my adsl connection.... ;) Thanks Enrico P. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto
2007 May 08
0
Beronet card - issue?
Hi all, I have a problem with my beronet card with 2 isdn. I think drivers and Asterisk are ok but the red led on the card always blinking. The card is connected with PBX. I post some conf: [root@gateway ~]# misdnportinfo Port 1: TE-mode BRI S/T interface line (for phone lines) -> Protocol: DSS1 (Euro ISDN) -> Layer 4 protocol 0x04000001 is detected, but not allowed for TE lib. ->
2006 Jun 14
0
Asterisk & wengophone
Hi I use Asterisk with some SIP phone (grandstrea), while with my notebook when I'm out of home/office I use X-lite and all work. Some days ago I try to install wengophone and I decided that I want replace X-lite for use wengophone as client for my Asterisk. One of the reasons is that wengophone support g729 codec. I make some test and I see that is possible to configure other sip server
2006 Oct 18
0
[OT] Nokia E60/61/70 and SIP
Martin Joseph wrote: > > > For all of us using these devices, I have some good news. There is a > self installable firmware update available from Nokia here (requires > windows box to install): > > http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate > > This seems to radically improve the behavior of the SIP client on my > E60. It seems to have
2007 Jan 10
1
Asterisk HA
Hi all, I have to make for a client an asterisk system for process up to 250 calls between conference and normal call. At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client require a failover system. Anyone have experience for this type of solution? Is better ultramonkey, dundi or SER proxy in front of * server? Thanks Enrico P.S. Now during all this year I have to work
2007 Feb 14
0
Asterisk & CME integration using h323
Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323. Cisco conf: dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten =>
2007 Jan 28
0
PHP sip client
Hi all, I want to write a simit sip client in PHP with asterisk API, in this moment I'm able to compose a number on my browser and call between 2 hw sip phone. I digit a number, my phone ring and after hanging up the cornet the second phone ring. But I want to add a features.... I want to hang up the cornet of my phone, compose the number in my browser and call a second phone. In witch
2007 Feb 21
0
IAX Realtime - show peers works?
hi all, I'm trying to set up some iax2 trunks in Realtime architecture with the same backend. All work better (make call, receive etc etc) but when I do "iax2 show peers" some asterisk don't show anything and other show the iax2 peers but with status "unknow". Name/Username Host Mask Port Status ctm1/trixbox 10.0.0.131 (S)
2007 Feb 09
0
*****SPAMZ***** Conference & Page question
Spam detection software, running on the system "placebo", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see enrico@pasqualotto.org for details. Content preview: Hi. I'm currently setting up a particular conference: 3 members
2013 Feb 21
2
Playback on h exten
Hi all, I'm trying to setup a Quiz/feedback for caller of call center when a agent hangup. I use Asterisk 1.8.16 and I'm trying with Queue and Dial with options c and g but every time I try to play something I got: -- Executing [301 at from-test:1] Dial("SIP/300-00000045", "SIP/301,60,rjtTg") in new stack -- Called SIP/301 -- SIP/301-00000046 is ringing
2007 Feb 05
0
*****SPAMZ***** Asterisk cluster - keep up connection?
Spam detection software, running on the system "placebo", has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see enrico@pasqualotto.org for details. Content preview: Hi all, how can I set up an asterisk cluster (using SER or hearbeat)
2006 Feb 27
3
Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug:
2007 Apr 17
4
Using meetme like call
hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because
2009 Apr 07
3
Logging Asterisk console
Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with "asterisk -rvvv". I need it in debugging purpose for tracking some bug. Thanks Enrico. -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type:
2009 Jan 21
0
Odp: R: Odp: Function able to identify the row and the column in a matrix
"enrico.foscolo2 at libero.it" <enrico.foscolo2 at libero.it> napsal dne 21.01.2009 11:54:49: > I can't use the function "outer" because my function fun() doesn't take vectors > as arguments. If you can not R-help at r-project.org mailing list https://stat.ethz.ch/mailman/listinfo/r-help PLEASE do read the posting guide