RE: [Asterisk-Users] VoiceMail
Check your voicemail.conf !! you can do your own date prompt (or no date prompt) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Waldo Rubinstein Envoye : lundi 20 juin 2005 15:56 A : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] VoiceMail I installed Asterisk Voicemail in an office and now most of the employees are complaining that when they're listening to the messages, it takes forever to listen to their messages. The reason being is that before the message is played, the voicemail says the full date and time when the message arrived and that takes a long time. It's like: Friday . June20th. 2000...and...5... etc (you get the idea). Is there anyway to shorten that or even give users the option to not play that? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] French SIP or IAX phones
Videotel !!! : French software, Video hard phone, Excellent browser... see it at : http://www.call.fr Works fine with Asterisk. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Martin Roy Envoye : vendredi 13 mai 2005 00:52 A : asterisk-users@lists.digium.com Objet : [Asterisk-Users] French SIP or IAX phones Is there any SIP or IAX phones that can be configure in french instead of english. I tested Cisco 7960 phones but I can't change the language it's only available in english with the SIP firmware. I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to french. Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] beginner in Asterisk configuration
Hello Sorry for english speaking peaple, but I just help this beginner in our natural language : French ;-) Je suis Français aussi, si tu as besoin d'un peu d'aide tu peux me joindre directement par mail Pour tester ta config : asterisk -gc Bonne chance -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Tutu Lord Envoyé : jeudi 12 mai 2005 09:58 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] beginner in Asterisk configuration hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _ MSN Hotmail : antivirus et antispam gratuits http://www.imagine-msn.com/hotmail/default.aspx?locale=fr-FR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice + Videosupport=yes - Fails!
Same problems as you... Eyebeam is not really fine in video... We have find some nasty bugs in it (PC freeze, codecs issues...) and no feedback from Xten after sending back reports (tcpdump and long descriptions). I think that EyeBeam works fine with... eyebeam. The software seems to be beta because of each version of Eyebeam I've download has differents bugs. Try with our hard-videophone ( ;-) ), Asterisk video features works. Perhaps a small problem in Intra Frame request (I've posted it in feature request without success). We will work on it ASAP. Nicolas http://www.call.fr -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Shadow Roldan Envoye : mardi 1 mars 2005 20:40 A : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [Asterisk-Users] Broadvoice + Videosupport=yes - Fails! Hi All First time poster, long time reader. I just ran into something really bizarre. I've enabled videosupport and have been testing sip calls with Xten Eyebeam software, it works (mostly) However, when I have Videosupport=yes In the [general] section of my sip.conf, broadvoice calls fail w/ We're sorry your call cannot be completed at this time So... I've commented it out and tried adding videosupport=yes to specific extensions, now video doesn't work as eyebeam reports remote user does not support video but broadvoice works. Bizarre I'm running CVS v1-0-02/15/05 Any ideas? _ Shadow Roldan IT Manager Zero G Software, Inc. tel: +1.415.512.7771 x 306 fax: +1.415.723.7244 mailto:[EMAIL PROTECTED] www.ZeroG.com The leading provider of multiplatform software deployment solutions. _ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling H323 channels with FC3 and RedhatSE3
Hello I'm trying for a while to compile and install OH323 channels on my two distribs... I have downloaded the src pwlib and h323 files versions given in the documentation. Make some RPMS with googled SPECs (and seems to give good results) Tried to compile the channels failed each time... (I have also tried with at-rpms oh323 and pwlib versions). Did someone who have already done the job could help me ? -I'm looking for working specs to compile pwlib and oh323- Thanks Nicolas. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF skipped when calling from ISDN to SIP...
Hello I have done the following test-network: IP-Phone = ASTERISK == ISDN PSTN Phone (SIP) + SER When I'm calling from the PSTN phone to the IP (SIP) phone: I cannot get ANY DTMF from PSTN, they seem destroyed by the codec (small scratches). I listen DTMF from IP-Phone (SIP INBAND!) When I'm calling from SIP phone to PSTN: Same result, no PSTN = IP DTMF ! Any ideas ? Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mysql and Voicemail
Hello try to enable mysql debug: log=/var/log/mysqlfull.log in your /etc/my.cnf and off course reload mysql then tail -f /var/log/mysqlfull.log it will show you if your asterisk connects to the DB... if not, it's a makefile problem... re-read tutorial... PS: don't forget to try if your full-log works by connect by anyways to the db. (And works fine like mysql ans show databases command) -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] la part de Alessio Focardi Envoye : mardi 28 decembre 2004 10:41 A : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Mysql and Voicemail Hi, I would like to enable mysql handling of voicemail boxes ... following that tutorial http://www.voip-info.org/wiki-Asterisk+voicemail+database so I modified the makefile of /apps directory to include USE_MYSQL_VM_INTERFACE=1 and copied mysql-vm-routines.h in the /apps dir, set up the db and some boxes in the table, also edited the voicemail.conf file. Everything compiles just fine, then when I start * I have no results, show voicemail users -- There are no voicemail users currently defined also if I try to check against a box with MailboxExists it does not result created Any idea of what I'm getting wrong ? tnx ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceMailMain login problem... new BUG ?
Hello
I think I've found a new bug, but first I'm asking for experts...
I have the following simple configuration :
in extensions.conf :
exten = 0660,1,VoicemailMain(${CALLERIDNUM})
So the caller is directly connected to his mailbox, it works great with
other users (like xlite, 0467161616, nfovdt...) but with the user pnunes :
When I tring to connect with the user pnunes I cannot enter into the
mailbox... it seems there is a mistake somewere (see the folder who is
nunespnunes instead of pnunes). Any idea ?
*CLI -- Executing VoiceMailMain(SIP/petitvillage-0813b1e0, pnunes)
in new stack
-- Playing 'vm-login' (language 'en')
-- No username but # key pressed. Using CID 'pnunes'
-- Playing 'vm-password' (language 'en')
-- Playing 'vm-youhave' (language 'en')
-- Playing 'vm-no' (language 'en')
-- Playing 'vm-messages' (language 'en')
-- Playing 'vm-opts' (language 'en')
-- Playing 'vm-helpexit' (language 'en')
-- Playing 'vm-options' (language 'en')
-- Recording the message
-- Playing 'vm-rec-busy' (language 'en')
-- Playing 'beep' (language 'en')
-- x=0, open writing: voicemail/default/nunespnunes/busy format: wav49,
0x80f0130
-- x=1, open writing: voicemail/default/nunespnunes/busy format: gsm,
0x80ed3b8
-- x=2, open writing: voicemail/default/nunespnunes/busy format: wav,
0x814ac70
PS: I'm using MYSQL Voicemail and the database seems correctly invoked :
SELECT password,fullname,email,pager,options FROM users WHERE
context='default' AND mailbox='pnunes'
Thanks for advice
Nicolas
http://www.call.fr
PS: We're planning making a small page on our VideoVoicemail test, it works
perfectly at this moment...
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[Asterisk-Users] Voicemail Video Attachement format
Hello I have tested the Video-Voicemail feature with our SIP Hard phone, it works great ! I'm trying to convert the h263 file (who cannot be played with an out of stock Windows Media Player) to another format for email forwarding (mpeg or another WMP recognised format) Anyone has tried to ? (I have tried transcode or ffmpeg without success) thanks for advice. Nicolas FOURNIL Nicolas P2P manager http://www.videotel.fr PS: We actualy doing a french translation for voicemail prompts, with also local changes (date format etc...) we will release it asap. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SJPhone SIP Tab
Hello I'm planning using meetme module with SIP videophones (Videotel Hard Phones), I have tried the meetme module with the v option but I haven't get any video... can someone get the video mode running in meetme ? Thanks a lot Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using meetme video mode with SIP ?
Hello I'm planning using meetme module with SIP videophones (Videotel Hard Phones), I have tried the meetme module with the v option but I haven't get any video... can someone get the video mode running in meetme ? Thanks a lot Nicolas SIP Hard VideoPhones http://www.call.fr ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
