similar to: SV: Re: CallerID Problem

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

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

Hi Chris, Yes the call was from PSTN and your solution worked great! I've read about SetCallerPres earlier but I didn't connect the dots this time. Thanks alot! :) Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Christoph F?rstaller Skickat: den 29 mars 2007 15:29 Till: Asterisk Users

Thanks! Do you have any suggestions on what I might do next. I have the phones, I have asterisk, and I have everything setup. But i can't get the login to work with the Polycom function. Nothing happens...and I can't find any readmes' or manuals. Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com

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

I don't quite follow you? There are USB headsets that don't require a soundcard at all. They have a built in soundcard which (I suppose) could be better than the crap they build into most laptops. 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:

>From what I understand it means that the *hardware* in your computer *acknowledges* the call as soon as it is recieved and then sends it to asterisk dialplan for processing. You would essentially need to put the delay before the call ever reaches asterisk. So this problem isn't asterisk related... if I've understood your question and the answer I found correctly. Regards, Jan

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

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

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

Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Campon, mini-queues, see asterisk tips and tricks on voipinfo... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zach A Sent: Monday, February 06, 2006 1:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help on queues I need practical examples showing

I'm having trouble setting callerid with teliax. I use a simple dial-out subroutine to set the callerid depending on the calling extension, and then dial out. Teliax is saying they're not seeing any callerid info. [DialOut] ; subroutine for dialing out. exten => s,1,NoOp(Context: DialOut called with outgoing number ${ARG1} ) exten => s,n,NoOp(${CALLERID(num)}XXXX) exten =>

Guys... Ive been having problems with my callerid and I have no more clues as to what I could be.. dates and times stamped on voicemail and info received on the phones display are off by +6 hours and also the date for example today is Jan 02 :) What can I do to modify this? __________________________________________________________________ Anton Krall

OK! I'm sorry about this. As I wrote earlier I'm totally lost... but I will try to explain the problem in steps bellow, ok. 1. I installed the rpm's for samba, Version 2.0.2a-ssl I think this is the version distributed with redhat linux 7.0 2. Then I changed the parameters in the /etc/samba/smb.conf file, and in this file I added the folowing parameters. [global] netbios name

Sorry about that, its just outlook that does that by default. But manually deleted your adress now in reply. I don't know what you mean with "top posting"? What I mean is that if you have another security on the connection (be it physical security - the connection doesn't go over public means, or VPN - connection level encryption) then you don't need another encryption on

Ok, I'll post all the messages on the screen. -----Ursprungligt meddelande----- Fr?n: H. Peter Anvin [mailto:hpa at zytor.com] Skickat: fr 2003-05-30 22:21 Till: Olsson Lars Kopia: syslinux at zytor.com ?mne: Re: [syslinux] Problems with PXELINUX 2.05pre1 Olsson Lars wrote: > Reading config based on hw address is really what we need, so we quickly > downloaded

Hi I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -----Ursprungligt

Hi Isn't this exactly what we _don't_ wanna do?! =) I suppose TDM and VoIP is supposed to interconnect not to be separated. i think it's nice with a busy list, it means some real hot stuff is happening, and that's good! rgds /staffan -----Ursprungligt meddelande----- Fr?n: Luciano Ramos [mailto:lramos@telviso.com.ar] Skickat: den 9 januari 2004 14:12 Till: