similar to: Asterisk Dial plan issue

Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090220/e2aa530c/attachment.htm

Hi all, I have the below peace of my AGI script...the problem here is that I cannot fetch the extension value to inside the script and assign it to another variable...I highlighted it in red #!/usr/bin/perl #use DBD::mysql; use DBI; use DBD::mysql; use Asterisk::AGI; ############################ #To read asterisk variable values. $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse();

Hi all, When trying to send a FAX I got the following error: Executing [003228949469 at micho:1] Dial("SIP/028949469-08466918", "SIP/ 003228949469 at 80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949469 -- Couldn't call 0032234534534 at 1.1.1.1.1 Where I should

We have an application for Asterisk that will require connecting 144 fax ports into the system. Faxes will route externally over a PRI. The 144 ports are for local fax machines within the building. Not all will be faxing simultaneously. We just need to be able to provide ports in the building to plug in lots of fax machines. The plan is to run an Asterisk server for about 100 phones and these

Dear All Can you please do me favor and let me know how can I stop my Asterisk ? Can you please confirm if the following procedure is correct to stop it ? #/etc/init.d/asterisk stop #cd /etc/init.d #chmod 0000 asterisk Let me thank you in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have 2 RHEL 5.4 systems where all facter lsb* items are missing. Any ideas? Sanitized facter output: architecture => i386 domain => missyou.edu facterversion => 1.5.7 fqdn => broken.missyou.edu hardwareisa => i686 hardwaremodel => i686 hostname => broken id => root interfaces => eth0,sit0 ipaddress => 1.2.1.5 ipaddress_eth0 => 1.2.1.5 is_virtual => true

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hello 'R' Users, I am very new on 'R', so excuse me if I ask something wrong. I have ASCII data and the colums of the data are looks like :- !------------------------- time,yr,mo,dy,hr,min,sec,lat,lon,ht,co2obs,sigma,co2model -- - -- !---------------------------- Each column has data value. Now I want to produce 2-d color maps, for example the plot should look like :- on

Hi, I'd like to reduce the size of the samba binaries. smbd for instance is about 1.5 meg, which is way to much if you ask me. Is there a way to create smaller binaries, maybe by compiling differently? thanx, Dennis ===== __________________________________________________ Do You Yahoo!? Yahoo! Greetings - Send FREE e-cards for every occasion! http://greetings.yahoo.com

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hello. I have succeffully open a memory buffer from a url-opus-file with (from Pascal fpc): HandleOP := op_test_memory(BufferURL[0],PipeBufferSize, Err); op_test_open(StreamIn[x].Data.HandleOP); I can retrieve the tags infos, all seems ok. But for reading, using this gives 0 outframes: outframes := op_read_float(Data.HandleOP, at Buffer[0], Wantframesm, nil); I suspect that some callback

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your recipient is using a codec that isn't ulaw or alaw). _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

adding gsm or just comment out the disallow and the 2 allows. (your = recipient is using a codec that isn't ulaw or alaw). =20 _____ =20 From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of michel = freiha Sent: Wednesday, January 28, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Dear all, i am using DGP 301 hard phone with my asterisk server. 1 : real time support is enabled .......all sip_buddies are stored in mysql database... 2: when i register my phone for first time it works fine.receives 2 ,3 calls then no call received.... hangup cause is congestion....i don't know why. 3: when i unregister or shutdown my dgp 301 hard phone .it still visible as

Hello All, now that the BUG 14343: (s3: vfs_full_audit: Add missing fcntl entry in vfs_op_names[] array.) was fixed I wanted to try 4.12 again. This is the relevant config which has been running for years: vfs objects = acl_xattr full_audit shadow_copy2 full_audit:failure = none full_audit:success = pwrite write rename unlink full_audit:prefix = IP=%I|USER=%u|MACHINE=%m|VOLUME=%S

Fred van Stappen wrote: > I have succeffully open a memory buffer from a url-opus-file with (from > Pascal fpc): > > HandleOP := op_test_memory(BufferURL[0],PipeBufferSize, Err); > op_test_open(StreamIn[x].Data.HandleOP); > > I can retrieve the tags infos, all seems ok. > > But for reading, using this gives 0 outframes: > > outframes := op_read_float(Data.HandleOP,