similar to: IVR/Voicemail, No Sound from Asterisk

We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks

Hi All - I pulled from a working system a TDM400 with one s110 fxs and three x100 fxos. I put it into a new box and the fxs no longer works. The fxos work just fine. I thought it was odd, but I chalked it up to a random chance failure and ordered another s110. The replacement doesn't work either, and now I'm confused. dahdi_scan recognizes the s110, but it says it fails. Output

Hi all, I am quite disappointed at the application AreskiCC. I have installed everything following the instructions but the thing doesn't want to work. First of all, when I start the index.php page, any name/password logs in. After the login it takes me to a page with a single option "LOGOUT" We are monitoring the database and it seems like the application doesn't

Hi, thanks for responding. Well, that's exactly wher I am stuck. My defines.php is with username=root and password=. but it still says the password is wrong. Now, let me be more specific: Postgres is running as user postgres at the Login: postgres, ok. The one thing I am not 100% sure is if php-pgsql is installed because I installed PHP from the Slackware 10.1 disk. Other then that, I

Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc.

Hello all, I am having a serious problem installing my * with a TDM04B. I made everything work, call are coming in and going out including using a GSM Box in channel Zap/2-1. I did setup voicemail like this on extensions.conf: [incoming] exten => s,1,Dial(SIP/2246,20) exten => s,2,Wait,2 exten => s,3,Voicemail(u${ME}) exten => s,4,Hangup exten => s,102,Wait,2 exten =>

Hi, I have installed AreskiCC on Slackware 10.1 with Asterisk latest CVS and Postgres 7.4. First of all the instructions are very confusing and hard to follow if you are not an expert. But, I managed to install it and.obviously t doesn't work. The other instructions I found on wiki are a great effort but incomplete. Basically the first thing that happens is that when I load

I've been using my Asterisk (At my house - 2 modem-type fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and I've been really happy with it, but I'm still having an echo problem that I've exhausted google with, and can't get straight... I think I've determined that because I'm using $7 voice modem clones for my FXOs that bad echo is going to

Hi, When trying to compile ASTCC i am getting the following error: root@asthome:/usr/src/astcc# make install mkdir -p /var/www mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Can't locate Asterisk/AGI.pm in @INC (@INC contains: /usr/lib/perl5/5.8.6/i486-linux /usr/lib/perl5/5.8.6

Hello everyone, I have Asterisk SVN-trunk-r7498 installed on a server (celeron 2.4 Ghz, 256MB) with a TDM40b a TDM04b and an avm fritz!card pci. I experience a problem with voicemail: my messages are good unless the incoming call comes from isdn, which means via the avm fritz!card. In this case (and in this case only) the message is disjointed and I can hear at most 1 second out of a 1 minute

I have a TMD400 card installed in a PC with one fxs (installed in slot 2) and two fxos (installed in slots 3 & 4).? fxos work fine but I am unable to get a dial tone for any devices connected to the fxs.? I have correctly connected the power supply to the card and I have even tried moving the card from slot 1 to 2 on the board. Below is from the console when I try to route a call from FXO on

AFAIK, this is not possible - but I'll throw it out there anyhow... I subscribe to telco voicemail, for the event that all my pstn lines are in use. Telco gives me a stutter-tone dialtone when I have a message waiting. Can a Zap card detect this stutter-tone and perform some action? I'm using TDM400P+FXOs and SIP devices. Thanks

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

Hi all, After finally making the web interface for AreskiCC work I am now running into new issues. 1 - In Asterisk the manager doesn't seem to connect 2 - When I try to create the file additional_areskicc_sip.conf it says "Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf' Any clues? -------------- next part -------------- An HTML

Hi I am fairly new to Asterisk. I will have few questions on one Asterisk system: Description: I looked at Asterisk Mall says that "The TrueLine SMB PBX is perfect for the small office providing service for up to 28 Telco lines". This is a 1U Rack Mount Single T1 Asterisk Appliance. Question: Does it mean it has 7 PCI bus and takes 7x4=28 channels (if they used 7 TDM400P card

Having ISDN FXOs and a bunch of ISDN DECT base stations, I went looking for a CAPI compatible active BRI adapter capable of NT mode. I have found that the 'Eicon Diva Server 4BRI' fulfills theses requirements. However, at EUR 1387 for four BRI interfaces, it does not come cheap. Are there other users of similar hardware that can point me to alternatives ?

Hi Boris, On Fri, Jul 28, 2023 at 02:26:12PM +0200, Boris Brezillon wrote: > On Fri, 28 Jul 2023 13:31:36 +0200 > Maxime Ripard <mripard at kernel.org> wrote: > > > Hi Danilo, > > > > On Thu, Jul 20, 2023 at 02:14:22AM +0200, Danilo Krummrich wrote: > > > Add infrastructure to keep track of GPU virtual address (VA) mappings > > > with a

On 20/07/2023 01:14, Danilo Krummrich wrote: > Add infrastructure to keep track of GPU virtual address (VA) mappings > with a decicated VA space manager implementation. > > New UAPIs, motivated by Vulkan sparse memory bindings graphics drivers > start implementing, allow userspace applications to request multiple and > arbitrary GPU VA mappings of buffer objects. The DRM GPU VA

Hi, We are using a samba server as a NTLM authenticator for our web filter. However, Samba closes the SMB connections after 60 seconds of inactivity, causing connections from the web filter to the Samba DC to get stuck in CLOSE_WAIT state, and failing to authenticate the users. Is the some way, other than changing IDLE_CLOSED_TIMEOUT, or (in Samba 3.0.23) commenting the check that closes the TCP

I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs). With Trixbox out of the mix and a regular phone connected I get the CID fine yet Trixbox shows 'unknown': dialparties.agi: Caller ID name is 'unknown' number is 'unknown' dialparties.agi: Methodology of ring is 'ringall' Here is my Zapata.conf if it helps: ############################# ; ; Zapata telephony