search for: zanin

Hi Everybody, as far as I see, I installed 1.2.12.1 on a 1.2.0 box. Everything ran OK, but VoiceMail application stated that there were no entries in voicemail.conf, so it didn't work. Installed again 1.2.0 and voil? the VoiceMail app. was working again. I asked to the group, but it seems I'm the only one with this issue! In Italy we say: "Chi lascia la via vecchia per la nuova, sa

Hi everybody, I have corrected this line in extensions.conf by stripping spaces off and now it executes: exten => 22999,1,VoiceMailMain(s${CALLERIDNUM}) when it runs, the mail box number is asked and password too. I expected no question were made, because I inserted CALLERIDNUMBER and s in front of box number. Anybody knows why? Thank to you all, very kind members of this list! Ciao Mauro

...ten => 22999,2,Wait(3) exten => 22999,3,Hangup Why do I get Forbidden 403 and one console display : Jul 25 09:48:09 WARNING[1117207472]: pbx.c:1274 pbx_extension_helper: No application 'VoiceMailMain ' for extension (home, 22999, 1) Anybody knows why? Ciao and thank you! Mauro Zanin

...------------------------------- In Italy plan for numbering assigns 3 as starting number for mobiles. 0 for ground lines. I'm able to call ordenary POTS and ISDN too. For mobiles seems to timeout, maybe gone to overlap mode? Anybody knows how to solve this issue? Thank you and ciao Mauro Zanin

Hi everybody I'm trying to make Asterisk 1.2.1 run under VMWARE and Suse 9.2. I use ZTDUMMY module for timing and ZTTEST gets an average precision of 98,4 %. Is there any way to improve it? Best regards Mauro Zanin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060117/945d544a/attachment.htm

Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro

...* works fine except for the ISDN CARD. It is always Channel D down, but if a Call comes in, it works perfectly for some time, both inbound and outbound. It prompts Channel D UP! If I disconnect the NT+ termination the Channel D goes down at once. Did I make something wrong? Best regards Mauro Zanin _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

...red led on back of card doesn't flash. Suse 9.2 has detected the card as a Tiger Jet card, since the chip on it is a Tiger 320. The second card configuration is still waiting for configuration, but I think this can be bad for Zaptel drivers. Has someone done something like this? Regards Mauro Zanin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050803/eb04cf46/attachment.htm

Hi everybody, I'm strugling between two devices: the both TE but one was set up as a TN. I have no current on that interface. I have tried to find some circuit over the net to power the connection, both commercial and home made. Can anybody give some hint? Ciao Mauro

Hi everybody, I have this issue: I need to automatically park an incoming call, play a welcome prompt and then connect to some extension but under extension user's command. I was thinking to use a small database to comunicate between asterisk and the main application. Has anybody had this kind of experience? Best regards Mauro

Hi everybody, I'm testing Asterisk 1.2.14(you can say: why such an old version? I say: I'm using Trixbox..). I must be as fast as a flash to press *2 and do an attended transfer. If I wait only a tenth of a second nothing happens. I think it is an issue. I have seen the source code and found nothing bad. Is this a known issue? Many thanks Best regards Mauro

Hi Everybody, I'm experiencing this kind of issue. One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel card. Everything seems to work but sometimes the third party caller when listening to MOH listens some "SSHHHHH!" instead of MOH, this is not continuos, MOH plays ok for, say, 20 seconds then the sound and then another 30 seconds of good MOH. We have some

I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start: Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15' Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered

Hi everybody, I'm new to this matter and I spent three days in trying to connect one SIP Softphone to an Asterisk Box. I always get error 401 or 403... I don't understand very well settings in Softphone program: con anybody show me how to set up a minimal running system with no public lines or external Proxies? Thank you for your kind help... Ciao Mauro

Hi everybody, I'm not able to make Asterisk startup by init.d in SUSE 9.2 Has anybody a sample of script to do this? Regards and thanks Mauro

Hi Atif make is a Unix's command which uses Makefile file for package's compilation. So after installing the complete development package from distribution disk, launch make. Ciao mauro

Hi * guys, I had a perfectly working * (1.2.0 version). I updated it to 1.2.12 and now VoiceMail app doesn't find entries in voicemail.conf any more. I recompiled only * 1.2.0 and installed it again and now Voicemail is up again, with no configuration's change! Anybody knows anything about this? Regards Mauro

Hi everybody, I have bought one 496 to use as ATA for two analog extensions in my office. I'm experiencing strange behaviuors. The ATA blocks itself and needs to be rebooted. Sometimes it hungs the lines(ISDN bristuffed HFC single isdn line). It was update to last available firmware. Has anybody those issues? Best regards Mauro