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Hi all, I have the below extension defined under sip.conf: [2203] type=friend username=2203 secret=123456 host=192.168.0.164 mailbox=2203 context=intern canreinvite=yes dtmfmode=rfc2833 When trying to register from a softphone installed on a PC behind a nat with IP=192.168.0.164, I got 503 FOrbidden...Does anyone have any idea about what could be the issue? Regards -------------- next part

Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Fred Posner fred at teamforrest.com Main: +1 (212) 937-7844 Direct: +1 (503) 914-0999 www.teamforrest.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081222/f431d833/attachment.htm

...nd point at 20 ms. I've seen discussions of setting this to a different size, but seems to be limited to the entire codec and not on a per peer basis. Anyone have luck with this? The Asterisk can be 1.4 or 1.6.x... I've a preference for 1.6.0.x but it's not set in stone :) Fred Posner fred at teamforrest.com

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

List, We're working on an upcoming job that may require us to access a web service (WS). I'm curious to hear peoples thoughts on the best way to do this with asterisk. We'll be submitting a single number to the WS and it will return a success or error. One solution would be to write a simple perl script to interface into to the WS, and use SYSTEM() from asterisk to call it.

Hi All; How to increase the waiting time between entering the digits for the analoge phone device that is connected to fxs? Is it by DigitTimeout? But how it will be apply for analoge station if the user just pickup the handset and dialed the number? Any help? Regards Bilal

Anyone know where I can get an incoming DID for Bogota, Colombia? Fred Posner fred at teamforrest.com Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 2162 bytes Desc: not available Url : http://lists.digium.com/pi...

On Fri, Jul 30, 2010 at 1:15 PM, Fred Posner <fred at teamforrest.com> wrote: > On Jul 30, 2010, at 5:04 AM, Andra=C5=BE wrote: > > > Ok, problem is another, when I run configure, it write this: > > checking for tds_version in -ltds... no > > configure: *** > > configure: *** The FreeTDS installation on th...

I missed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa

Hi All, how are you? I would like to know from you if the problem can be below is a BUG of the asterisk-1.4.21. I did an upgrade version of asterisk-1.2.18 for the version of asterisk-1.4.21 and now, when users try to sip friend outgoing calls through Polycom IP 330 appliances can not be the traditional way or with the telephone handset in his hand and digit dialing digit to receive the following

Hi every one, I am using asterisk version 1.6.2...... i did not install mysql data base and when i tried to register a client from SIPp xml file..... it is registered.... My questions are 1. where can i find that registered client? 2. when i type the command "core stop now" it exists and the registered users are not shown why this is happening? 3. Is it compulsary

Hello Everyone, I have an installation where the client has a Microsoft SQL database that holds all of their case information. They would like the asterisk system to require users to enter a valid case number when making an outgoing call. I?m seeing some documentation regarding people using Microsoft SQL for CDR storage, however nothing regarding validating authentication using a Microsoft SQL

This is a small 12 line system, internal extensions 150 - 180. I didn't have a phone on 151. Here's the sip.conf stanza: ;;[151] ;;type=friend ;;context=longdistance ;;callerid="Conf Room" <151> ;;secret=0000 ;;host=dynamic ;;qualify=yes ;;dtmfmode=rfc2833 ;;allow=all ;;defaultuser=151 ;;nat=yes ;;canreinvite=no There's no DISA. And then somehow (how???) ip address

Hi, Inspired by a recent rant about one particular provider, I am getting very curious about something I've never mastered. I'd like someone to explain this here or at least post a link or two that can educate me and probably countless others who have no knowledge in this area. I'm sure there are several of you reading this that know all about the subject. What are the various

I have a straightforward application of ddply() and summarize(): ddply(MyFrame, .(Treatment, Week), summarize, MeanValue=mean(MyVar)) This works just fine: Treatment Week MeanValue 1 MyDrug BASELINE 5.91 2 MyDrug WEEK 1 4.68 3 MyDrug WEEK 2 4.08 4 MyDrug WEEK 3 3.67 5 MyDrug WEEK 4 2.96 6 MyDrug WEEK 5 2.57 7 MyDrug

I got an error when trying to extract a 1-column subset of a data frame (called "my.output") created by dplyr/summarize. The ncol() function says that my.output has 4 columns, but "my.output[4]" fails. Note that converting my.output using as.data.frame() makes for a happy ending. Is this the intended behavior of dplyr? Tx, John > library(dplyr) > # set up data frame