similar to: Dial and option G

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

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)

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

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

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

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

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

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. ->

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 --

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]:

Hi all, I have a question on asterisk conference. Now I use appl Meetme with option A & x because when a marked person hangup I want to close all the conference. But what I have to do if I want two marked person and kill the conference when one of two hangup? Is possible? Thanks. Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org

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

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 =>

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

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

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:

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)

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

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

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags