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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

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 =

Class: default File: /var/lib/asterisk/moh//reno_project-system File: /var/lib/asterisk/moh//macroform-robot_dity File: /var/lib/asterisk/moh//manolo_camp-morning_coffee File: /var/lib/asterisk/moh//macroform-cold_day File: /var/lib/asterisk/moh//macroform-the_simplicity Class: none File: /var/lib/asterisk/moh/.nomusic_reserved/silence

2013/2/12 Doug Lytle <support at drdos.info> > >> non-standard locations such as /usr/local/sbin > > If compiling from source, it'd normally be specified by the --prefix > option: > > ./configure --prefix=/usr/local > > Doug > > -- > Ben Franklin quote: > > "Those who would give up Essential Liberty to purchase a little Temporary >

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, 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, "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 *

Server on the other sites, and get a response. However, no client has resolve the names of the Servers. I thought I could get around this by adding the following line into my smb.conf: name resolve order = hosts lmhosts wins bcast (though I'm sure there are better ways of achieving this!!) But this doesn't solve it! My Samba Server includes a WIN Server and a DHCP Server, which issues

games is pretty much in its infancy at this point so I hope to see that grow in the future. My first question as I looked around was wondering whether gaming apps that were in there had been checked were checked in Crossover Games or Crossover Office? Maybe I missed it. I don't know yet. Anyway, I am really happy to see a second path for Windows gamers in Linux. That's not really me

IDAMP cache somehow ends up with an unmapped SID2UID entry (i.e value = -1) and the SID2GID entry expires. At this stage winbindd returns unmapped for a SID-to-UNIX-IDs request. This results in smbd giving incorrect group memberships and incorrect resource access, until the SID2UID entry expires. This is surely a bug? Samba version is 3.6.6 as bundled with our current Solaris 10 patch revision

event - which it rejects. The voicemail notifications ARE working on the device. Any way to get rid of this message (while keeping the MWI on the phone)? ------=_NextPart_000_0072_01C950DB.BAD4D4E0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" =

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, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect

Thanks for the reply! I tried the instructions on Frank's IE5.5 page, including copying all the files and directories to my ~/.wine/fake_windows/ directory (except jsnl.dll, which I could not find...) I do not know if dcom95 exists somewhere in one of the directories that I copied across, but I know that exactly the same thing happens when I try to run IE on the Windows partition (instead of