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...doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:voicemail@mydomain.com Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:.

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

...ing calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/c131f743/attachment.htm

...from the update the uniqueid field into mysql, is not written and ASTPP do not charge the calls. I got an eye to cdr_mysql.c and I found that at line 212, into one insert query, uniqueid is missing. But I can be wrong. In any case, somebody got same problem? Any suggestions? Thanks to all. -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060305/daa06dd5/attachment.htm

...ctober 10th, but nothing changed. These are my user settings: ---- [221] type=friend username=221 secret=secret host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes context=local mailbox=221@local callerid="221" <221> accountcode=221 qualify=yes ---- Any ideas? Thanks to all. -- .:FaberK:.

Why don't u attach the setup page of the phone ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di FaberK Inviato: gioved? 13 ottobre 2005 17.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] PA168S/AT320P Right now, but nothing changed. 2005/10/13, Kanuri, Seshu (Company IT) <Seshu.Kanuri@morganstanley.com>: > have you configured the STUN server on...

...t_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 What's wrong? Thanks -- .:FaberK:.

.... My problem is that instead of ANSWERED or something like, into the CDR database records, I find only numbers. This is also a problem to let ASTPP works, infact I receive an error: ERROR - ERROR - ERROR - ERROR - ERROR DISPOSITION NOT MATCHED and the call has no cost. Any suggestions? Thanks -- .:FaberK:.

...r the voice that say the charge per minute?(I've payed the '5' euro, and from that moment I've got it!). Of course I understand that is to let me know how much I'm going to spend, but I do not like it, expecially when I'm with clients. Any links, suggestions? Thanks -- .:FaberK:.

Hi all, I've got an Asterisk box with 1 Sangoma A102 and 1 Ericsson PBX. I need to use Asterisk as E1 line for the Ericsson PBX. How do I have to connect them? I'm trying to connect the Sangoma to the Ericsson, but RED alarms remain. Any suggestions? Thanks -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060125/511eafbf/attachment.htm

Hi to all, I've just started to see that Asterisk-gui from Digium. Does anybody know, when the first official-realese will be released? Thanks to all -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071012/09197260/attachment.htm

...e in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: XXX.XXX.XXX.XXX Now I've checked into the router, and the VAD was already unset. Using normal IP-telephones, everything is perfect. Does anyone, got an idea or already got problems with that router? Thanks to all -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060228/24705641/attachment.htm

...maybe you?ve got the answer...! When a caller(not internal, but from PSTN) call *, I need to let him hear a message, before * answer and the bill start running. If is not clear, just let me know. caller->telco(telco bill to the caller as soon as * answer)->asterisk Thanks in advance. -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060320/75aa4548/attachment.htm

...gure: *** The Radius Client installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** without explicitly specifying --with-radius But the installation of radiusclient, didn't give me any problems. Any hints? Thanks -- .:FaberK:.

...n numbers and 4 concurrent calls) and 15 SIP extensions. The receptionist has a SNOM-360. How many SIP accounts would you configure on that phone? Only one would be enough? One SIP account, has a limit on concurrent calls? I saw that the SNOM-360 can handle up to eleven SIP accounts. Thanks -- .:FaberK:.

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

...em up right now. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ------------------------------ Message: 7 Date: Thu, 19 May 2005 23:57:45 +0800 From: VoIP Newbie <voip.newbie@gmail.com> Subject: Re: [Asterisk-Users] asterisk-oh323 build problems To: FaberK <f.faberk@gmail.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <62b5865d05051908576d725acc@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Read README file first. You will get a clue. On 5/19/05, FaberK...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can the Dial() command tell the difference between busy and congestion? At the moment it seems to be treating them both the same on my server. I want to route the calls out via a SIP gateway unless that is congested, in which case dial out through my POTS line (using an X100P). It seems a bit pointless to try dialling the POTS line when the SIP

...t_val_t h323_clear_call(const char*)': wrapper.cxx:1230: warning: unused variable `ClearCallThread*clearCallThread' make[1]: *** [wrapper.o] Error 1 make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 What's wrong? Thanks -- .:FaberK:.

...called my number and I've picked up the phone while is was "ringing", and the call was not there... I'm using port 5061 and rtp port 8000. Username and password are correct. I'm sure because I've setup a voip phone with those and it worked. Any ideas? Thanks a lot -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050904/71429389/attachment.htm