similar to: Status of conference calls at Astricon ?

Displaying 20 results from an estimated 2000 matches similar to: "Status of conference calls at Astricon ?"

2003 Apr 18
1
Account code on SIP
I was wondering if the accountcode flag works with sip channels. I was looking into the debug and ,even if I have the line accountcode=XXX into the users sections of my sip.conf, I don't see it logged into the cdr. Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso
2003 Feb 21
0
I4l outgoing dtmf problem.
Hi. I'm working with i4l with asterisk CVS-02/21/03-13:59:12, plus i4l (chan modem i4l *dsp patched* and kernel 2.4.19 patched to disable dtmf). All seems ok (apart some echo issues that seems gone with mec2 aggressive suppressor), but outgoing dtmf doesn't work . or at least I hear the very first part of the dtmf, but then it seems suppressed. here's my modem.conf [interfaces]
2003 Mar 03
0
Asterisk log rotation
Hi. Has anyone provided an easy way to rotate asterisk log files into /var/log/asterisk. I want to do that, because I prefer to have full logging enabled in the debug file and the messages file, but could became pretty big. Same apply for cdr-csv files. I wanted to setup a logrotate rule, but was thinking if I must use a kill -HUP to asterisk. (never tried HUP with asterisk... don't know if
2003 Apr 02
0
Zap flash bug?
Hi. I'm experiencing that bug with flash on zaptel. That's the problem: Zap/A call Zap/B Zap/B flash transfers to Zap/C Now Zap/A is online with Zap/C Till now all ok... but now if Zap/C wants to transfer again, it can't... the debug says that it got a WinkFlash when call not up or ringing (as attached below, Zap/10 is Zap/C in my example) Apr 2 09:14:01 DEBUG[32789]: File
2003 Apr 11
1
Strange Sip problem?
Hi. I'm getting a strange sip issue, with latest cvs. I was tring the *8 extension for call pickup on sip, but I forget to define the callgroup & pickupgroup in sip.conf . Now when I dial *8 from the crisco phone and hangup, the channel in asterisk don't go down and I'm not able to dial from the phone again. If I do a softhangup on the rem. console it does nothing and the
2004 Apr 15
3
* Announcement * Astricon 2004 - call for speakers!
We're proud to announce Astricon 2004 - the first Asterisk user's and developer's conference! * Where? Atlanta, USA * When? September 22-24, 2004 The conference is arranged in partnership with Digium.inc and the keynote speaker is Mark Spencer, lead developer of Asterisk - the Open Source PBX. Among the speakers already signed on are Ed Guy of Pulver.com, John Todd, Jeremy McNamara
2003 Jul 18
0
FW: Sip codec preferences
Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:
2003 Jul 23
2
SIP info
I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause
2004 Apr 16
1
Windows Drivers for Wildcard FXO Card
And if you want to use it with windows telephony software, such as answering machine or modem communications software, you can probably take the drivers for the Intel MD3200 based modem, modify the .inf for the Digium vendor and device ID. I have not tried this, but since the MD3200 modem works that way in Linux, the X100P may work that way in Windows. Then you would have a $100 winmodem! Let
2003 Jul 16
0
Sip codec preferences
Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones. I have 2 1 sip phone that's outside in the "world", and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729
2004 Jul 30
5
Non standard usage of X100P card.
I have two X100P card in my box. I want to connect regular phone (not the phone line!) to one of thse cards. Does anybody think about the same? I don't really want an expensive solution buying additional card with FXS port, I prefer to make something by myself. It'll be great if somebody can point me to technical materials or show electric scheme of such converter. I believe it should
2004 Jan 13
2
Nufone.net net wackiness?
I can't send mail to any addresses in nufone.net; they all get rejected by a spam blocker. And their website is gone, too!! The URL leads to a "parking site." My accounts still seem to work, but I wonder WTH is going on? Thx. B.
2004 Jul 29
2
Astricon Conference Call?
I thought the interesting exercise would be to use asterisk for the task. Couldn't we use a kind of distributed conference call where a few key select/high bandwidth asterisk servers form the main conference and then have multiple layers of conferencing. That way no one server/network is saturated. -----Original Message----- From: asterisk-users-admin@lists.digium.com
2005 Mar 10
6
NuFone
Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm
2004 Apr 29
1
Stop thinking - just do it! *** Speak at Astricon 2004!
Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal
2005 May 24
3
PHPAGI problems
Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (gsm|ilbc|speex), > priority = mine -- Executing
2004 Jul 26
6
Can't dial SIP<->EuroISDN (HFC-S based PCI ISDN card): Unable to create channel of type 'Zap'
Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the
2005 Jan 08
3
ASTCC questions
Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have
2004 Jun 16
1
IAX registration
Hi, I have a nufone connection (IAX2), works fine. In my iax.conf I do not specify a time interval that * needs to renew registrations with nufone server. However I can see following registration messages on my cli every 90 seconds (approximately) --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569 --Registered to '198.22.67.70', who sees us as 69.5X.XXX.XXX:4569
2003 Sep 28
0
Asterisk CVS viewer on line
Hi to all. I've put on line a cvs viewer for asterisk source code. Is based onto the suite horde+chora. The website is http://asterisk.espia-net.net The cvs modules shown are * asterisk * asterisk-addons * zaptel * zapata * libpri * libr2 * libiax * libiax2 * gnophone * phpconfig * gastman all revisions, branch , comments & whatever cvs is has been preserved. this could be a sort of