similar to: Voicemal error

hi, i've a question. is it possible to buy digium x100p card from italy in some store (also online) without ordering it from USA? on more, did anyone buy a modem with intel chipset 537 or md3200 and where (in italy)? Thanks __________________________________________________________________ Tiscali ADSL Senza Canone, paga solo quello che consumi! Non perdere la promozione valida fino al 27

Hi, I'd like to use PSTN and analogic telephone with a Asterisk server which works on a LAN connected to the outside with fiber optic, on which voice stream passes too. Do I need some particular solution, some particular card to make it works or I just need a Digium or similar fxo card? Thanks, Bob __________________________________________________________________ Tiscali ADSL Senza Canone,

Hi, I've a question. I need a fxo card to use with *, but i saw on site that also modem with chipset intel works. My question is: are there other chipset (beyond to written ones), intel or not, that works? If not, has anyone from Italy bought a modem with chipset intel 537 or md3200? because i don't succeed to find it!! Thanx, Bob

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Hi everybody, just to confirm that I understood it right (and that the info isn't obsolete): I have to store the voicemail audio data in an external mysql DB. In http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage I read that this is only possible via ODBC and *NOT* via native mySQL (like with CDR storage). I would like to avoid using ODBC. So: Is thsi still correct? Will

-----Original Message----- From: Adams, Gavin Sent: Monday, August 04, 2003 6:10 PM To: 'asterisk-users@lists.digium.com' Subject: Cisco 7960, SIP, NAT, Voicemal Hey all, I've got a couple 79xx phones working peer-to-peer and am now trying to work on the voice mail. In extensions.conf: [ATL] exten => 4001,1,Dial(SIP/gadams)|10 exten => 4001,2,Voicemail,u4001 exten =>

I forgot to take out the portion that would read in the voicemail boxes from the text file. If you want to leave it in then you could have some voicemail boxes defined in the text voicemail.conf. I do not, so I have removed it. Below is the new patch: *** app_voicemail.c 2004-06-23 07:55:54.000000000 -0600 --- app_voicemail.c.new 2004-06-23 07:55:47.000000000 -0600 *************** *** 49,61 ****

Hello all, I am just getting going on building my system, but I thought I'd send you all a patch that I wrote so the command: show voicemail users issued from the CLI works properly when there is a postgres backend for the voicemail. The current version of the app does not display the voicemail boxes found in a database. It is called in the load_config function. I haven't done

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hi ALL, When i was accessing the voice message it suddenly goes dead and after that i couldn't able to retrieve the voicemessage again from my mailbox . This happens once in a while for any configured mailboxes I am using the following system configuration. asterisk 1.4.22.1 odbc storage of voicemail messages centos 5.2 64bit unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2

I am currently having issues when trying to change my password in Voicemail. I am not utilizing realtime because I believe there is a problem with MWI being sent to phones and realtime databases (this may have changed since I last used Asterisk). Whenever I try to change the password for the account, the asterisk program gives a Sig fault (signal 11). I do see the voicemail.conf.new file

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback

I have been testing with asterisk 1.4.11 and have found a segmentation fault while using voicemail. It happens when I try to forward a voicemail. As soon as I press the option the server crashes. I ran asterisk up inside gdb and got the following stack trace ==================================================================== Program received signal SIGSEGV, Segmentation fault. 0x00140adf in

The comment says `Works in 2.3.0: BDR fixed inline [xy]axp' and 2.2.1 is not 2.3.0. Your example is not reproducible, so we cannot check. If you get a copy of R-devel (as the FAQ and posting guide ask you to before posting) you can check for yourself. Please do study the FAQ and avoid time-wasting. On Sat, 28 Jan 2006 cain at u.washington.edu wrote: > Full_Name: Kevin Cain >

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Hi, Is the following behaviour a bug or a feature ? Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces : [Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:457

In Asterisk 1.2.13 in app/app_voicemail.c, line 4700 ext_pass_cmd is checked to decide whether to use vm_change_password or vm_change_password_shell to change a user's password for his voicemail account. I wonder, what is the difference between vm_change_password and vm_change_password_shell - what is that shell? The only reference I found on the Internet was the following bug report: