search for: sarin

...sprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r BJ Weschke Skickat: den 23 februari 2006 13:44 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: [Asterisk-Users] Polycom 501 ACDlogin On 2/23/06, jan.sarin@securia.se <jan.sarin@securia.se> wrote: > Hi, > > I have several Polycom 501 connected to asterisk. The phone has an > ACD-login function that I'd like to use. But I can't find find much > information about this. > > I've read a post on bugs@digium > (ht...

...e? Attribute Values Default Interpretation call.rejectBusyOnDnd 0, 1 1 If set to 1, reject all incoming calls with the reason "busy" if do-not-disturb is enabled. Have not used, but looks like it may ignore the key if this is 0 Let us know... On May 4, 2006, at 2:22 AM, <jan.sarin@securia.se> <jan.sarin@securia.se> wrote: Hi, Is it possible to disable the DND feature on a Polycom 501? Regards, Jan _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update op...

Hi, I have several Polycom 501 connected to asterisk. The phone has an ACD-login function that I'd like to use. But I can't find find much information about this. I've read a post on bugs@digium (http://bugs.digium.com/view.php?id=6119) about this function but I'm not really clear on if this is actually working or not? Has anyone actually used the Polycom ACD-login function

Hi, What USB headset would you recomend? We have some laptop soundcards that are really bad and I would be glad if you could share your experiences when changing to a USB headset instead of using the built in soundcard in your computer. Thanks! Regards, Jan

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

Hi, Is it possible to keep asterisk from logging exactly everything? I can do the logger rotate and keep the files small enough, but I think it's unneccesary to log exactly all data. File grows by about 5 gb per month! Thanks! Regards, Jan

Hi, Is it possible to configure asterisk so it doesn't play MoH from certain phones? Regards, Jan

Hi, Lately we've noticed that some Zap channels on one of our PRIs are unavailable. We have 2 PRI lines with 60 channels in total. On the first PRI there are currently 20 channels that are not being used for some reason. I tried googling around and found some similar problems but there really was no solution (?). I'm not sure if this problem has occured now because of more load on the

Im curious. Does anyone have experienced echo-problems that later where solved by buying a hardware-echo canceller such as the Wildcard TE411P? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r James Harper Skickat: den 6 februari 2006 11:46 Till: Asterisk Users Mailing List - Non-Commercial

Hi, How do I read (make sense of) the timestamp in the queue_log? I'm probably just slow but I don't understand it. Thanks! Regards, Jan

...s. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r El Flynn Skickat: den 24 maj 2006 10:17 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: [Asterisk-Users] USB headsets? jan.sarin@securia.se wrote: > > We have some laptop soundcards that are really bad and I would be glad > if you could share your experiences when changing to a USB headset > instead of using the built in soundcard in your computer. > Well, IMO if the soundcards are already crap to start o...

...nd go back to the main menu instead of hanging up and redialing, or how can a queue be started for an extension, i.e. if 3-4 callers dial 201 and 201 is busy, instead of sending calls to voice mails, start a queue and let them wait in queue. Zeeshan A Zakaria -----Original Message----- From: jan.sarin@securia.se [mailto:jan.sarin@securia.se] Sent: Monday, February 06, 2006 12:52 PM To: asterisk-users@lists.digium.com Subject: SV: [Asterisk-Users] Help on queues What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [m...

What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've

...lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Brian J. Murrell Skickat: den 2 februari 2006 22:37 Till: asterisk-users@lists.digium.com ?mne: Re: SV: [Asterisk-Users] delaying "answer" for a number of ringsor anamount of time On Thu, 2006-02-02 at 22:08 +0100, jan.sarin@securia.se wrote: > http://lists.digium.com/pipermail/asterisk-users/2005-September/125146 > .html OK. The hardware is a wildcard though. How does that answer apply? Isn't it asterisk itself that is picking that call up? Can't it delay the pick up? Maybe I am just misunderstandin...

Hi, I have a problem with setting outgoing caller id to "nothing" (secret) on our Wildcard TE405P connected to a swedish euroisdn line. Caller ID seems to work fine when connecting the same line to a Ericsson PBX - so something must be wrong in my settings, but I don't know what. I've tried: exten => _*70X.,1,Set(CALLERID(name)="") exten =>

This is what i found on Cisco's site: "Symptoms: Media negotiation fails for SIP calls and the terminating gateway replies with a "488" message to an Invite message. Conditions: This symptom is observed on a Cisco platform when the terminating gateway is configured with the G279B (annex B) codec and when the Session Description Protocol (SDP) for the incoming Invite message

Thanks for the quick response! I read about logrotate at voip-info.org but I didn't quite understand it. I'm no asterisk/linux expert unfortunately. First of all. What exactly does happen when I run: /usr/sbin/asterisk -rx 'logger rotate' Does it clear the file and create a new one? Can I run this manually without any interruption in the system? And what does the script do? I

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...) > 22. Re: PRI Card (Jared Smith) > 23. Re: Sip Providers (Al Bochter) > 24. Re: how to use call transfer (Bruno De Luca) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 19 Jul 2007 16:11:15 +0200 >From: <jan.sarin at securia.se> >Subject: Re: [asterisk-users] Zap channels unavailable? >To: <asterisk-users at lists.digium.com> >Message-ID: > <0FF4F1903968F943B5EA2521CD5296C16EEF36 at exchange.securia.local> >Content-Type: text/plain; charset="us-ascii" > >Hi, >...

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time I want Asterisk to delay answering the POTS line via a