similar to: Poopy errors on quad wcfxo

Displaying 20 results from an estimated 3000 matches similar to: "Poopy errors on quad wcfxo"

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 Nov 24
0
Picking an open channel (FXO port) for outbo und calls
Thanks to everyone for your quick responses to this question. I'm very excited about the Asterisk project, and the growing community seems to be very active these days. Hopefully when the time comes for our county's transition to VoIP we may be able to go for an Asterisk-based solution. -- Tony Kava Network Administrator Pottawattamie County, Iowa -----Original Message----- From:
2003 Apr 08
1
Wiki for the * community.
Hi 2 all. I was thinking to start a little web site with phpwiki, to let the * community build a sort of shared documentation 'bout * & related. That because in a wiki "place" all grows faster, and is also the right place to share experiences. For example it's right to have documentation about * installations, ie who has done what with asterisk Till now we don't know
2003 Jun 13
3
Call queues for phone operator
Hi. I was wondering how can I make incoming calls to wait if the phone operator is busy. I've 8 incoming lines, with 30 extensions. What I need is if the operator is busy with call nr #1 , the new incoming call waits until the op. is free. Looking into app_queue seems the way to go. So I want to ask if I'm right or wrong: I set up only a queue , is to say operatorq, where the only member
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
2003 Oct 07
1
[PATCH] allow announcements in app_dial
Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch
2004 Jan 31
1
asterisk php status viewer
since I was annoyed this morning, I wrote this simple php script to output channel status from asterisk manager. <disclaimer> that's very bad written, nor commented... I wrote that just for fun </disclaimer> and if someone will use that / improve it , just lemme know. http://asterisk.espia-net.net (wrote with php 4.3.3 and depends on Event: StatusComplete, so a recent * cvs
2003 Oct 19
0
Flastman 0.0.1-pre-alpha
Hi. My first 'snapshot' of flastman is out. Flastman stands for FLash ASTerisk MANager. written in flash, this first version is just a proof of concept, ie doesn't nothing except for logging in/out & displaying manager events while logged in. But is realtime & in any flash-enabled browser. Not very useful yet, but I'm going to improve it. For the hardcore testers, grab it
2003 Nov 14
0
Hidden bug in *8 call pickup with Sip
Hi. Seems that there's a hidden bug in sip call pickup with *8. If I pickup a Sip ringing phone from another sip phone on the very first ring (before the first ring completes), the ringing phone doesn't stop ringing, however the call is connected ok. But, if I pickup the call after the first ring, (after the first, the second, or during the second, or other silly combinations always
2004 Jan 20
0
[A-bit-OT] Power Over Ethernet Discovery process
Hi, Since someone asked, here's how POE standard does discovery process for a POE device. of course is a passive detection... but that's why you don't have POE always-on on a POE enabled switch port.... you can find more info in article area of http://www.poweroverethernet.com and full specs @ http://www.ieee802.org/3/af/index.html You will find a resistance value in the quote
2004 Apr 26
0
Re: [Asterisk-cvs] asterisk BUGS,1.7.2.1,1.7.2.2
Hi, about select vs poll differences, like in cvs message below <snip> > > Modified Files: > Tag: v1-0_stable > BUGS > Log Message: > Update 1.0 BUGS file to be aware of select vs. poll stuff <snip> > +* The number of channels that can be simultaneously run through Asterisk > + may be more limited than in development releases due to the use of > +
2004 May 28
0
E1 channel bank problem
Hi all. I have and E1 channel bank from Loop Telecom. there's a little issue with it, I cannot ring the phones on fxs interface, but can connect without issue them. What happens: I dial the phone on port 1, asterisk says "Zap/1 is ringing", but the phone on the analog port doesn't ring. but if I take off hook the ringed phone, asterisk detects the answer at they're bridged
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:
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 May 25
2
Message Waiting and VoiceMail 2
Hi. I noticed that if new messages are recorded with voicemail2 , they're not detected by the message waiting indicator, so the mailbox=XXXX param has no effect, and no message waiting is sent to the phone (sip & adsi, or stutter dialtone) Any hint? -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmgi Srl
2003 May 01
2
Max number of connection in IAX ?
Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo
2003 Sep 17
1
Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >