similar to: Creating new voicemail accounts

Hey all, I just tried running a 'make rpm' on a fresh install of Fedora Core 4 and ran into an error near the end of the build process. This is the output of the build when the error occurs: done rm -f /tmp/asterisk/var/lib/asterisk/mohmp3/sample-hold.mp3 mkdir -p /tmp/asterisk/var/spool/asterisk/voicemail/default/1234/INBOX :>

Hi Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

Hello, I have moved from Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k (debian pkg) to CVS-HEAD, and realtime. Compiled no problem and now running, with realtime extensions and sip users in postgres (ODBC connection) database, trunking also works. I have looked on google, wiki, and this mailing list, along with talking to some peers, but to no avail. My problem revolves around voicemail. I have looked

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

Hi there. Is there a way I can tell asterisk to play *vm-isunavail* (person unavailable) instead of *vm-isonphone* (person is on the phone) Many thanks, Pierre -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070111/d75f1d29/attachment.htm

Hi, the app HasNewVoiceMail can't find my voicemail. This is the errormessage: Dec 3 14:24:01 NOTICE[12222481]: app_hasnewvoicemail.c:104 hasvoicemail_exec: Voice mailbox 25 at /var/spool/asterisk/voicemail/default/25/(null) does not exist however this is the output of lspbx:~# ls -l /var/spool/asterisk/voicemail/default/25/ total 316 -rwx------ 1 root root 11814 2003-11-22 18:18

Hi all, 2 days ago, managed to install rhat and asterisk, starting with 2 sip phones, simple config. CLI reports : "playing 'vm-theperson' (language 'no')" when transferred to voicemail after timeout, or "playing 'vm-password' (language 'no')" when dialing into voicemail extension. But, no sound, nothing heard in phones. First time, first

I reported this earlier, and have got reports that others have the same promlem on UNIX machines, so it is not only a windows problem S I file a bug report. The function (boot.stat) given at the end produces a postscript file, which cannot be included correctly in LaTeX. Specifically, the image in LaTeX (when translated by dvips to postscript) becomes very small, not using the bounding box, and

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi Samba Team I have a urgent problem. I have a Unix box that's running Samba Server, I can see the Server on the NT network but when I try to access the share I get the message: "\\Samba is not accessible". The network path was not found. On the WINS Server I have a static mapping to the IP and I'm able to do a ping from the Unix and to The NT network and from the NT box

911 Help! I accidentially deleted all directories under /var/spool/asterisk I did use the backup facility not too long ago but cannot find the process for restore. However, I don't believe a full restore is needed -- I just need to know the names of the directories under /var/spool/asterisk and re-create them (I hope!). Can some kind soul give me some direction or tell me the directory

I have quite an old version of Chameleon Mail, currently the prompts played when leaving a message are ? -- Executing VoiceMail("SIP/209-3b0e", "u5") in new stack -- Playing 'vm-theperson' (language 'en') -- Playing 'digits/5' (language 'en') -- Playing 'vm-isunavail' (language 'en') -- Playing

Hello! I've seen, some people experienced this problem earlyer, but I have not found any solution in the mailing lists... So the problem is, copying files from an NT Workstation (SP5 ) on a Samba server running under Linux (2.0.10) in some (rare) cases results in files exactly of the right size but filled with NULL bytes. I'm using samba 2.0.5a and likely since version 2.0.0 this

Hi Is there a phone comparison matrix I could consult I have a series of features that I would like to evaluate on the most common phones on the market example: dual-ethernet POE / direct power / both number of lines speed dials programmable buttons BLF LEDS Headset plug conference call built in hands free operation display size codecs communication protocol (SIP, h.323) price availability

I am using the following macro to dial a ZAP channel. When I dial in, * answers and I go to voicemail. I never hear any ringing, though. It doesn't work with the Ringing command before or after the Dial command. [macro-zapdial] ; ; call a ZAP extension for ${ARG2} seconds, and then voice mail ; ${ARG1} - Extension ; ${ARG2} - Time to ring exten => s,1,Dial(ZAP/${ARG1},${ARG2}) exten

Hi I've suddenly started encountering a strange issue. Sometimes, when a call is made into our system, an extension answered the phone but I can see no mention of it being bridged in the console. Also, the server does not seem to think that it is answered and then goes to voicemail. We are using asterisk 1.4.17 Here is the console output for one of these calls, it was me ringing a

Hello All, I am using Asterisk 12 and sipml5 as front-end and when i call from one to another the call will ring on other end but when i allow the camera access call will terminated automatically. I have attached the logs of Asterisk, if some one will get something useful Please reply on the same. Thanks and Regards, Anant == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5

Anyone wanting to go ahead and use VoiceMail2 will probably need this new version of the addmailbox utility.. I have updated the addmailbox utility to create the correct directory structure required by VM2 and copy the required files to the correct locations.. I just called it addmailbox2.. I don't know if it will be added to the CVS that is up to Mark.. Remember to make it executable.. :)