similar to: Astreicon presentations

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 folks, Is there anywhere I can possibly get a model of the exam itself, maybe possible scenarios for the prac, etc? To people who have done the exam....any helpful hints ? Thanks,

Hi guys, Having issue with getting CDR to write to MS-SQL via ODBC. > cdr_odbc: Connected to freetds-connector > cdr_odbc: Error in PREPARE -1 > cdr_odbc: Query FAILED Call not logged! == Spawn extension (cisco, ##########, 2) exited non-zero on 'IAX2/astYYYY-507 Isql test: [xxx at YYYY asterisk]# isql freetds-connector XXXXXXX YYYYYYYYY

Hi guys, Having issues with doing an insert statement using ast 1.4.24: [START] dsn=mssql-asterisk write=INSERT INTO testdb (callarrival,callerid) VALUES ('${VAL1}','${VAL2}') SET(ODBC_START()${TIMESTAMP},${CALLERID(num)}) No errors pop up on execute, but nothing gets inserted. Read and update work fine, Wondering where I'm going off track with this, Thanks,

Hi, I have used R rmgarch package to implement EGARCH ADCC model from which I can extract conditional covariance matrix. Now I would like to introduce positive and/or negative shocks to see the asymmetric response of covariance. I have come to know that impulse response function (IRF) or volatility IRF is not compatible for any asymmetric models, therefore, the only way to introduce shocks into

Hi folks, How would I go about running a stored procedure call from asterisk via func_odbc. I'm after an example entry in func_odbc if possible for ast 1.4 Also, if someone could post an insert statement that actually works, would be nice. Thanks, :)

Hello all, Let me try this again with some code. I am trying to make a new object and it's not working. I get : Error in rematchDefinition(definition, fdef, mnames, fnames, signature) : methods can add arguments to the generic only if '...' is an argument to the generic as my error. I have included my code. It is from the xcms package. Any help would be great even if

I saw several threads requesting an Asterisk forum to complement the email list. i.e. http://lists.digium.com/pipermail/asterisk-dev/2004-February/003103.html I recently created an Asterisk forum within TMC's popular VoIP forums for everyone to use. http://voip-forum.tmcnet.com/voip-forum/forum/forum_topics.asp?FID=15

Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? Thanks, Daniel

Hello. I am new to this forum. If this issue has been discussed already, please inform me, but my STFW have indicated that it has not. If SS7 were available on * would it make commercial sense to use it? There are case studies on the web (courtesy of the Death Star :->) for ISPs with 10,000+ subscriber lines showing clear cost savings in transitioning from PRI to SS7 controlled IMT. My

Does the application PGSQL has been removed from Asterisk? Couldn't find it on Asterisk source and addons. Atenciosamente, Vin?cius Fontes Gerente de Seguran?a da Informa??o Canall Tecnologia em Comunica??es Passo Fundo - RS - Brasil +55 54 2104-7000 Information Security Manager Canall Tecnologia em Comunica??es Passo Fundo - RS - Brazil +55 54 2104-7000

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

I know Vonage doesn't officially have a "bring your own device" type program, but they do offer a softphone. Has anyone gotten Asterisk to connect directly to Vonage? This would be a great help!! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/5bc9f644/attachment.htm

This is a fairly large list of hardware phones, features, and URLs, with some vendors I've never heard of before on it. I suspect that any phone that doesn't explicitly say "SIP" is an H.323 phone. http://www.tmcnet.com/it/0403/0403sg.htm#sidebar2 JT

Asterisk receives TMC Labs Internet Telephony Innovation Award http://www.tmcnet.com/it/0704/tmclabs.htm Jim James H. Thompson jht@lava.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040708/133dbbde/attachment.htm

I am using asterisk 1.6.1.6 and have been setting up a system to use a Postgresql database as the realtime DB via the ODBC route. I have got extensions and voicemail working but am having trouble with SIP The problem seems to be with using a schema. If I put the table "sip" in the schema "foo" then I add this entry to extconfig.conf sippeers => odbc,psqldb,foo.sip Restart

Hi all, I'm trying to set up an odbc connection to a ms-sql server from an asterisk 1.6.1 install My problem is that I cannot get asterisk to build func_odbc & res_odbc.so I installed yum -y install unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel And then went on to reconfigure / recompile asterisk after a ./configure --with-odbc=/usr/lib/ I get

Hi All, Is there a way to change the mappings of disconnect reasons to certain SIP messages? E.G. I need to change the mapping for SIP 402 ?Payment Required? from 16 (normal termination) like it is in 1.4.24 to 21 (call rejected) as defined in RFC 3398. For me this is a big issue because my dial plan will look for alternative termination in the event of network error (e.g. reason 3 or 21 which is

Hoping someone can help me understand what is happening here; we start asterisk as a service at boot (actually, with heartbeat) on CentOS using the asterisk init script installed with "make config" upon reboot of the server (when the asterisk service is first started by heartbeat) we get color in the console when we connect to it using asterisk -r after the execution of

Hi, When I was testing an IVR, I realized I miss a function I would call GotoIfTimeWithOffset. Today, this IVR is using function AEL GotoIfTime in several places. The problem is if it's 11pm at the moment I'm testing this IVR, I can't nicely test the 9am or 2pm branch. GotoIfTimeWithOffset would get 2 incoming arguments : - the first is a time range (just like GotoIfTime), - the