similar to: Problem after upgrade to 1.2.7.1

Hi, i've downloaded/compiled/installed the bristuffed asterisk Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a and i'm using it with an hfc card. It runs on a debian 3.1 sarge machine with kernel 2.6.11. Asterisk works well if i configure the card using isdn4linux. I'm having problems dialing out (not tried the input yet). This is the output from asterisk: -- Accepting AUTHENTICATED call from

I have the same problem in a box with 2 HFC-PCI, but i already remove the row in modprobe.conf and load the module manually. Both cards works fine Any idea ? Giordano -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Julian J. M. Inviato: luned? 27 giugno 2005 12.04 A:

Hello everybody. I have a problem with my Billion ISDN card. When I run Asterisk (asterisk -vvvc) on five minutes (aprox.) it puts in the screen this: zaphfc: empty HDLC frame or bad CRC received (framelen = 3, stat = 0xff, card = 0). in the framelen it change 3 and 2. Anyone knows something about it? Thanks a lot. bye! -------------- next part -------------- An HTML attachment was

Hi, I'm setting up an asterisk server with this hardware configuration: AMD Athlon 1000 Mhz 256 MB ram 3ware ATA raid controller 2 * Ethernet controller 2 * ISDN HFC controller One ethernet controller is connected directly to the internet (public IP) One ethernet controller is connected to the internal lan One ISDN controller is connected to the public telephone network One ISDN controller

Add a new constant ETH_P_802_3_MIN, the minimum ethernet type for an 802.3 frame. Frames with a lower value in the ethernet type field are Ethernet II. Also update all the users of this value that David Miller and I could find to use the new constant. Also correct a bug in util.c. The comparison with ETH_P_802_3_MIN should be >= not >. As suggested by Jesse Gross. Compile tested only.

Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some "numbers" and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal

Hi, Does anyone have some experience with Nokia 32 Terminal (it is an analog GSM Gateway)? After a configuration I can make only incoming calls, I'm not able to do any outgoing. Nokia signalize an error (4 short tones), when I try to phone someone. I tried postpaid simcards as well as prepaid simcards with the same result. Does anyone try to connect this gateway to Asterisk PBX if so what

Hi, I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Maybe there are some other non commercial applications which work under

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Hi, I've been struggling with this for a quite long time. Maybe I am not the first asterisk user with this problem, (I try to search on google, but I didn't find anything good). My point is: I try to set up * to work as a fax server. Each incoming fax (from PSTN) should be received on email. Luckily it works. I didn't notice any problem with receiving faxes on email, so rxfax works

Hi All, Can anyone tell me what are the <dev1> <dev2> parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM

Hi, I want to extend my asterisk stuff and buy some Planet devices, to be certain I'm going to buy PLANET VIP-050 with FXO and FXS modules. Has anyone heard about it. Is it compatible with Asterisk, or it would cause a lot of problems. Dose anyone have some experience with it?? All the best Andrutto ---------------------------------------------------------------------- Oferty sprzedazy

Hi! After upgrading to asterisk 1.2.7.1 (from 1.2.4) it seems that app_rxfax doesnt work any more. I've installed spandsp-0.0.2pre25 (the same problem with spandsp-0.0.3pre6.tgz ) app_rxfax.c, app_txfax.c and made the Makefile patch from the same directory. When starting asterisk I always get the follwing error message: [app_rxfax.so]Apr 14 18:54:20 WARNING[7223]: loader.c:325

Hi, I was using Asterisk with Hylafax via IAX Modem. It works fine until I upgraded to Asterisk 1.2.7.1 I didn't change any configuration but it seems that Asterisk does not get the call from IAXModem anymore. I'm doing something like this Asterisk <--> IAXModem <--> Hylafax Usually when I use sendfax -n -d 260XXX somefile I'll see Asterisk receiving the call in

The make seems to go okay. [root@somebox asterisk-1.2.7.1]# uname -a Linux somebox.org 2.6.16-1.2080_FC5smp #1 SMP i686 i686 i386 GNU/Linux mkdir -p /var/lib/asterisk/sounds/digits mkdir -p /var/lib/asterisk/sounds/priv-callerintros for x in sounds/digits/*.gsm; do \ if grep -q "^%`basename $x`%" sounds.txt; then \ install -m 644 $x

(This is a shameless copy-paste from the note I posted on http://bugs.digium.com/view.php?id=5090) I have again backported the whole T.38 shebang to the stable branch. The port was based on two versions of the t38passthrough branch: r19125, the latest unconflicted automerge, and r13623, the latest version without the new chan_sip flag structure. Basically, the port contains everything that

Has anyone managed to add the bristuff patch to 1.2.7.1 successfully? My attempts has ended up bad, so if anyone has a working patchfile for 1.2.7.1 I would be grateful to receive it. Thanks, Vidar -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060430/1c56d94e/attachment.htm

I've got my res_mysql.conf stating: [general] dbhost = 127.0.0.1 dbname = switchref dbuser = asteriskuser dbpass = xxxxxxx dbport = 3306 and my extconfig.conf stating: sipusers => mysql,switchref,sip_buddies sippeers => mysql,switchref,sip_buddies When Asterisk starts, and I show peers and show users, I don't see anything that is in the database. When looking at the traffic

Hi list! Are there any changes in the behaviour of the Dial command between 1.2.7.1 and 1.2.8.? I am forwarding calls to my legacy PBX using : exten => s,1,Dial(Zap/g1/8210,90,r) Ever since I upgraded to 1.2.9 it seems as if the Legacy PBX is no longer receiving the extension I am calling on the PBX and the call gets dropped to the switchboard extension on the legacy PBX. Did I goof up