similar to: Why does Digium not respect their own development guidelines?

Digium has discontinued their ATA iaxy adapter; don't blame them, too expensive so they can not compete. The adapter is upgraded automaticaly when it is connected to new asterisk version; since this adapter is discontinued will it still work with asterisk 1.6 and beyond or will it be\ just a "door stopper"? -- Joseph

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 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 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 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

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

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

Dear all, I have sent this problem earlier but maybe its not detail, here I try to write more detail. I hope anybody can help me to point out the problem. First of all I used Ubuntu 12.04 x64 both for domain0 and domainU with modification to run under xen hypervisor and work with remus. I follow and configured the remus with this notes

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we

I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100305/b92821c0/attachment.htm

Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since

I've noticed a bug with even recent OpenSSH products, where if the host disconnects during a certain period of time, the connection becomes frozen causing possible expolit problems . For example [root at portal ~] users root [root at portal ~] uptime -u (used to show how many users the box believes is logged on) 2 Users [root at portal ~] In theory this trapped connection can and has

Hi Folks, Are all the astricon presentations up? I'm especially after the one that tilghman did. I caught the tail end of the prez when I decided to skip the session I was attending and go for that one. :)

Hello List: I'm working on a funny scenario, where I'm bouncing calls from a Cisco call center into asterisk. Cisco call center has some logic that if a customer calls in, an agent is logged into a given extension... if Cisco sends a customer call to that extension, and there is a ring with no answer after a preset amount of time, Cisco concludes the agent is unavailable, kicks the agent

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

Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten => 1003,1,Wait(1) exten => 1003,n,Playtones(!1050/1000) exten => 1003,n,Wait(1) exten => 1003,n,StopPlaytones exten => 1003,n,Echo exten => 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

Hi, I had the following odd behaviour in Asterisk 1.2 - We are migrating to 1.6, and I will re-test ASAP, though it is quite hard to replicate, but I am curious to know whether it is a known IAX issue in 1.2. We had 2 users in iax.conf: [user1] username=user1 secret=secret1 context=context1 host=iax.hostname.com [user2] username=user2 secret= context=context2 host=dynamic deny=0.0.0.0/0.0.0.0

Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Thanks -- - Eric Smith