similar to: issues.asterisk.org

I am trying to connect two machines to each other with an T1 crossover cable. The first machine has two TE120P cards - one connecting to the telco on an ISDN PRI. The second to a crossover T1 cable to a second machine which has one TE120P card. Telco <-cA-> Machine1 <-cB-> Machine2 Machine1: Two TE120P cards Machine2: One TE120P card cA: Standard T1 Cable cB: Crossover T1

Anyone know if the MySQL() application has a configurable timeout? If it tries to connect to a bogus IP, it's timeout seems to be a few minutes. I'd like to cut it down to a few seconds. Doug. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part --------------

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,

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, 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

Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_item&px=OTY0OQ Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

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

>>Your membership in the mailing list asterisk-users has been disabled >>due to excessive bounces The last bounce received from you was dated Anybody else seeing this? My mail server logs don't show any issues. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety."

Hi Friends, i have installed and configure asterisk-1.8.0. When i have tried asterisk start get below errors and not able to start asterisk. *FD 32767 exceeds the maximum size of ast_fdset!* Thanks in advance. -- Best Regards, Rajnikant Vanza -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

We currently have an Asterisk system that is hooked up to our old paging speakers via sound card, plugged into two amps. Each amp drives up to 8 analog speakers in each warehouse (we have 2). Both warehouses are around 30k square feet. Both have a large number of printing presses. The computer system is that is running Asterisk is around 10 years old and starting to fail. I'm looking to

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

Last month I moved a 1.4.x Asterisk install to Asterisk 11.5.1. Everything is working well, until I noticed that Caller ID between facilities are showing properly, on the phone display, until the handset is picked up, then it's displaying NAME NOT FOUND. I do database lookup against extension number (Of remote PBX) and use that as the MySQL key to pull name. Snippet below: -- Executing

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

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

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 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