similar to: How to install in /usr/local/sbin instead of /usr/sbin ? [SOLVED]

Hi, I am working on vxml to record voice. I have trouble with getting url of recorded voice. I want to save and I am using java to get record parameter from url and it returns a string which is audio/basic:len(123123):p0x5a6e6241, but I want to get file object or base64 string with parameter or to relate returning string with path in asterisk server, are there any way to do this? --

All, I'm trying to send an SMS directly from asterisk but it doesn't seem to be working. The SMS() function does create an outgoing file but doesn't deliver the SMS. Can anyone help me to understand how SMS() works. Thanks. extensions.conf example: same => n,SMS(hello,a,17654307001,"hello nick") - nick

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl =

Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten => *100*,1,AGI(test_app.pl) ... exten => 190,1,Answer() ... exten => *100*,1,AGI(never_reached.pl) ... A "normal dialplan reload command" would echo no warning or

Hi, Am testing out the iCal functionality but when changing the URL am faced with the following warning: [Feb 25 20:55:20] WARNING[6234] res_calendar_icalendar.c: Unable to retrieve iCalendar 'dummycal' from 'https://webmail.domain.com/home/user at domain.com/Calendar/': Server certificate verification failed: certificate issued for a different hostname, issuer is not trusted

I'm now getting these errors: [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP even

Hi, "Asterisk 1.8.10.0-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2012-03-08 23:05:09 UTC" We have some problem when running a AGI script (build with PHP), existing channels (all of them) gets a "hickup" and then continues. We are using AGI to lookup incoming calls in directory. It is kinda annoying, and I don't understand how it can be

Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1

Hi every body; I want to intall some softwars working with my Asterisk server and I get these erreurs : * error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. error: cannot seek `/dev/sda'. /usr/sbin/grub-probe: error: cannot seek `/dev/sda'. dpkg: error processing grub-pc (--configure): subprocess installed post-installation script returned error exit status 1 *

I'm hoping someone can help me here. I've purchased replacement systems for 3 aging 1.4.x installs. I'm hoping to setup Asterisk 11, dahdi 2.6.1 and Oslec. I'm also moving those installs from Mandriva 10.0 to Debian 6.06 (Squeeze). In my testing, the TE220P PCIe cards that I have, the timing was awful on both slots, so I compiled Kernel 3.6.9 from kernel.org. Timing jumped to

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

not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to make call file using php command line..but when executing php from AGI, it is not working..kindly see the attachment if bellow text is not readable...___________________________________________________ File: /etc/asterisk/extensions.conf[call]exten => call,1,Answerexten => call,n,Playback(hello-world)exten =>

Hello, On a LAN, is it possible to install a bluetooth dongle on one workstation (at this time, this workstation OS is not specified) and use it with chan_mobile ? I've read some USB over IP (or Ethernet) middleware exist but I'm not certain I'm looking at the right direction. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I am going to register asterisk sip users through active directory accounts LDAP (that is a separated server with ip : 192.168.11.17) So I have followed the below link as well: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html

hi folks. recently some of our customers complained about bad voice quality on the phone system. i looked at the logs and found a lot of these: [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort (6) on D-channel of span 1 [2012-11-03 08:26:54] NOTICE[11305]

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

am I missing something crucial? -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248

So I'm working with our Digium D40's and we're not using DPMA. This video ( http://www.youtube.com/watch?v=zcuocp01pfM#t=35s ) shows presence information being displayed in the Contacts application. Obviously the video is showing DPMA in play. Is it possible to enable this functionality without it? Is this status information only available on higher-end Digium phones? In the contacts

> > ---------- Forwarded message ---------- > From: Doug Lytle <support@drdos.info> > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Date: Tue, 10 Oct 2006 16:25:11 -0400 > Subject: Re: [asterisk-users] How big is *your* dialplan?? > Steve Murphy wrote: > > Hello! > > > > In my relentless