similar to: Got SIP response 481 "Invalid CSeq Number" backfrom

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,

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

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

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

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

Just use a cisco with 5 T1 ports and have everything over IP use ultra monkey to load balance your asterisk boxes. I have found this works very well. .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

What can you develop in? What are you comfortable? I use PHP for testing then convert into C shared objects. ..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 Asterisk Sent:

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

If you were at Astricon 2004, Astricon Madrid, Astricon 2005 or ANY other astricon event and you have pictures we would love to have them. Please drop us a email and we will make arrangements to get your pictures from you. We are located at: http://astri2005.netdr.biz Brian Fertig This email was scanned by: Mcafee GroupShield ---------------- CONFIDENTIAL DISCLAMER

FYI Brian Fertig Treasury disconnects tax on long-distance calls WASHINGTON (MarketWatch) - The brief Spanish-American War ended more than a century ago, but not the federal tax assessed to fund the victory. Until now. On Thursday, the U.S. Treasury said it would stop collecting the 3% federal excise tax on long-distance calls, a fee originally assessed in 1898. The government also said it

I am having a slight issue. I am trying to register 2 asterisk boxes with GNUGK and when I try to add the 2nd it gets denied cause of it saying its a duplicate. How do I change the configs to allow more than one asterisk box register to the same GK? brian This email was scanned by: Mcafee GroupShield ---------------- CONFIDENTIAL DISCLAMER ---------------- All information provided in this

Mitel ICP 3300 & Asterisk, Is possible that integration? (C?sar) -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Jueves, 25 de Mayo de 2006 03:00 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 22, Issue 147 Send

Well I know from personal experience that NuFone is working on a solution for its customers as fast as it can. I know they found an alternate termination provider and are working to have a solution for the TF and Local DID's he currently has on his platform. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

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

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

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

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

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 --------------

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

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