similar to: Hidden bug in *8 call pickup with 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

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:

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

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

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

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]

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

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

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

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

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

Anyone having problems getting transfer to work here is the answer... It appears the manual is incorrect.. The manual says: 1)Press "Transfer" button. 2)Dial the target extension. 3)Hangup the phone. This will disconnect the call.. Here is how it can be done.. Matteo gave this solution.. (thanks) NOTE:'Use # as Dial Key:' must be set to YES To trasnfer: 1)Press

Hi all, I'm experiencing problems with the TDM card with 4 fxo modules. on all tests, if the cards has 4 modules, I get "poopy" kernel messages on the card. The card works for sometime,then hangs and a asterisk restart must be done, along with kern modules unload/reload . if I remove the first module, the card works without problems at all on the remaining 3 modules. using latest

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

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 > +

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

Hi. Has anyone experienced with the new firmware .77 ? There's Day Light Saving time now, but haven't time to play with it, till now. Matteo. -- Matteo Brancaleoni Espia System Administrator - IT services Website : http://www.espia.it Email : mbrancaleoni@espia.it

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

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:

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