similar to: Help with TNT and Asterisk

I created some scripts to logrotate. I am having a problem. After I do it, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL

Hello all, We are looking for some hardware requirements/recommendations to be able to handle a full DS3's worth of TDM -> VoIP traffic. The DS3 would bring 24 calls per T1 x 28 T1s = 672 simultaneous calls. We would then need to convert those calls into G729 SIP VoIP calls to send to our asterisk box over ethernet. Since everything is going in/out of asterisk is 729, and no features

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Hi, can anyone who has the Areski Calling Card solution on Asterisk working comment on it? Is is stable enough for a production system? Any pros and cons? thx, Arnd

Could someone explain to a non-US dummy the following phrases I have seen on the list. "I can provide you with tier 1 termination 6/6. I can blend or NPANXX breakout." "We provide US48 termination, blended rate for 1 MOU and above is .008 with 6/6." What is 6/6? What is US48? What is blended? What is MOU? What is NPANXX breakout? -------------- next part --------------

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

Running fc3 with current cvs-head... Is there a nice way to rotate the /var/log/asterisk/messages file without shutting down asterisk? I'm currently rotating the log files via cron, however my script requires asterisk to be shut down, which also kills any outstanding cli sessions (eg, asterisk -rvvvvv). Would like to rotate the files without killing the cli session. Any reasonable way to

Most gateways I have found are only sold overseas. Do these work in the US? My provider is T-Mobile (using their Blackberries). They support: GSM (I am pretty sure) GPRS EDGE We get unlimited Cell to Cell minutes and would like to leverage the possible savings. Does anyone know of a product that they have been happy with? SIP or Analog is fine although SIP (or IAX) is preferred for the

I am about to build a Dual Opteron Asterisk box as our soon to be production server. Is Core 4 supported or should I stay with Core 3? There was a recent post about an issue with the latest Core 3 Kernel and zaptel. I had the same experience, but just rolled back to the previous version of the Kernel on Core 3 on our evaluation server. Thanks in advance

Seems to be all morning. I have not been able to access for several hours now. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 7:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi

Is it possible to act as a middle man on a T1 line? My installation currently has an aging Inter-Tel Axxess box with a T1 coming in (16 in, 8 out). Rather than adding and replacing phones and cards as they die, I would like to slowly migrate to a asterisk SIP installation. I want to take the incoming T1 line, use any available outgoing lines for outgoing SIP, intercept any incoming lines and

I am running a prepaid application with Asterisk. When authentication has to be done by DTMF everything works fine. However when the user is authenticated directly from the sip phone, the channel variables seems to disappear. Trying to retrieve the channel status always returns -1 instead of the 6 that happens normally. It also seems to affected the DIALSTATUS and ANSWEREDTIME variables. The

Hi List, I'm wondering if someone who uses VoipJet as their termination service would do me a favor. If I call the American Airlines reservation number (1-800-433-7300), the call gets connected, but after 30 seconds asterisk drops the call responding that no one answered. I'm using areskicc2 (calling card app) as an authentication system and I don't know if that is what is

I have put a pbx into a resort with Polyycom phones, everythign works great, except the kitchen staff cannot hear the phone ring. I know many legacy systems employ a big red flashing light, any ideas on doing something similar? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: netconcepts_anguilla@yahoo.com

Did someone monitor the PRI's or trunks some way? I tried with MRTG and Andrea Fino module but it never worked for me. Any other experience? I want to track the use of my PRI's and trunks using graphical as MRTG does each 5 minute, day, week & Year. But the option of the 5 Minutes I don't think is usefull, We need something more realtime. Thanks, Carlos Alperin

Hi, list! I'm playing on an Asterisk@home installation, since a month or two. I've had no trouble setting it up 'n running. I've bought a couple of DLINK's DPH-140S SIP Phones, to use with Asterisk. >From this phones, I can make & receive calls with no trouble, but, when I try to use some "interactive" function (eg Directory or Voicemail), the phone seems

are you giving answer()? ..o-------------------------------------------------------o.. Brian Fertig Network/Systems Engineer IT Administrator Planet Telecom, Inc. Tampa,FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose Comellas Sent: Friday, September 30, 2005 10:32 AM To: Asterisk Users