similar to: Upgrade issues

Hi All, I have been struggling with the auto dial out in asterisk. I am trying to get a call to be auto dialed and play back a message once the line is answered. So far I have been unsuccessful. Currently what happens is I have my .call file. I mv it into /var/spool/asterisk/outgoing. The call is initiated and that all works, my problem is that it does not wait for the line to be

Hi all, I am having an issue with a multiple ISP setup. I have followed the docs online and I think I have everything setup correctly but I can get the desired traffice to go out my secondary ISP. A quick run down on what I am trying to acomplish. I want to send all sip/iax traffic out one ISP in the net zone and then send all other traffic out my secondary ISP in the dsl zone. Attached is

Hello, I am currently using MusicOnHold(mpg123), and it works just fine, but every once in a while I will get a flurry of warnings in the CLI like those below and Asterisk will freeze completely, and the only way to come out of it is with a kill -9 . Is mpg123 causing my problem? Is there a specific format of MP3 that should be used/avoided to not have errors like these? Any help would be greatly

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

Hello group I just update to the newest CVS now I'm not able to get asterisk to start. No error during the make or make install I did a make clean before the make;make install Any help would be great!!!! Here is the output asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding realtime_ext to mysql/realtime/extensions_table == Binding

Hello, I've done a very straightforward install of Asterisk, and can't seem to get it started. This is a proof-of-concept installation, and currently does not have any T1/E1 or FXO/FXS cards in it. I just want to use it as an internal SIP server for now. However, when I try to start Asterisk, it dies with the following messages: Junk at the beginning 49443303 Warning, flexibel

Hi All, Does anyone know of a good captive portal that I can get working with shorewall? I am trying to find one that will authenticate users and if a user does not have an account they can create one and pay for service temporarily. Like a hot spot. I''ve been looking a Wifidog but I''m not sure how to get it working with shorewall and also it just creates free accounts, so I

Looking at the error I tried moving chan_modem* out of the modules folder and asterisk started and its working again... Not sure what changed in the chan_modem_i4l.so but removing it from the folder fixed my problem. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Hall Sent: Sunday, January 23, 2005

Hi all, I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545 zt_pri_error: PRI: !! Got a UA, but i'm in state 1

Hi,all I am newer to Asterisk.My Asterisk version is the newest CVS-HEAD.now something appears in the console CLI like below these,I don't know what's happen to my Asterisk Server.Could anybody help me? Thanks Junk at the beginning Warning, flexibel rate not heavily tested! Junk at the beginning Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested!

Hello, Hope this isn't TOO much of a newb question... I just created a new WBEL server with a fresh install of asterisk. When I try to load asterisk, it dies with some cryptic error messages. I've googled for them, but haven't found anything helpful. If anyone can point me in the right direction, I'd much appreciate it... Below is the output when I try to load asterisk.

I am using CVS-v1-0-04/20/05-12:31:26. Windows Messenger5.1 works MOH fine. After I stop MOH on Windows Messenger, if the hungup signal could not send to *, the sip channel(e.g.,SIP/52001-08ca) for this MOH remains. Then the user trys again MOH, a new sip channel starts. And again the hugup signal can not send to *,......... When I 'stop now' from CLI> , * cleanups the remaining sip

Hi. I am having the following warning when using music on hold. It works from X-Lite to Grandstream. I get a lot of errors and warnings. 1.Warning, flexibel rate not heavily tested! 2. NOTICE[1100258240]: res_musiconhold.c:260 monmp3thread: Request to schedule in the past?!?! Thanks for any help. Full Output below: Jan 30 10:24:55 WARNING[1133718080]: chan_sip.c:486

Hello. Is this normal. Or does it mean there is a problem ? ------------------------- stop now Beginning asterisk shutdown.... Executing last minute cleanups == Destroying any remaining musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Yuck! Error in buffer handling...: Broken pipe Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0).

Hi, I have a data-matrix: > PID sato hrs fim health 214 3 4.376430 6.582958 5 193 6 4.361825 3.138525 6 8441 6 4.205771 3.835886 7 7525 6 4.284489 3.245139 6 6806 7 4.168926 2.821833 7 5682 7 1.788707 1.212653 7 5225 6 1.651463 1.436980 7 4845 6 1.692710 1.267359 4 4552 5 1.686448 1.220539 6

Hi. I am testing E-mail delivery by using dovecot 1.1 alpha6. But, The following message(syslog) is output while receiving mail. And, I'm try to looked for this reason in that source code. Hum... I think why the deliver is not using a 'MAILBOX_TRANSACTION_FLAG_ASSIGN_UIDS' flag. # deliver/deliver.c:142::deliver_save() function - calling mailbox_transactio n_begin() The

Hi Please advice me about the following problem. * Problem find_by_xxx method does not work correctly the following caes with sybase adaptor. * How to reproducible model :users t.column :code, :string <-- :string column t.column :last_name, :string t.column :first_name,:string >> User.find(1) => #<User:0x2abe232a68e8

Hi, Probably my problem can be helped by very specific people. Anyway... I am debuting with Windows NT (portuguese language version) and Samba and experiencing problems because I unsucceed to map Samba to Windows with the message "the account is not authorized to login from this workstation" (this is my translation from portuguese). The TCPIP connection sound good as long as I can

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

Hi ! :) Does someone have an idea of the reason why my MusicOnHold (with clean mp3 downloaded from http://aussievoip.com.au/wiki-MOH) is always having interruptions and micro-cuts ? No high load of the system, no swap, no hard disk r/w, mpg123 seems running well... nothing ! Except a little message at startup : "Warning, flexibel rate not heavily tested!" I'm getting