similar to: Assigning a higher irq to a digium card

Hi, I have some experience in sending SMSs using smsclient. I call the german Vodafone SMSC (01722278020), and smsclient takes approx 20 secs to send a SMS. The hardware is an Sedlbauer ISDN card. Now, I want to do the same using asterisk and a digium PRI card. I dialed using the manager with: action: originate channel: Zap/g4/01722278020 ... I assumed, the call will fail, because the remote

Hi, can someone sched a light what exactly mean the read write permissions in manager.conf? [public] secret = private deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Lets say I want some users to use dial through manager interface. But don't want to allow them to run asterisk commands?

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

Hello Although not stictly a asterisk issue, any help would be apreciated. Firstly a few notes on the snom 360, which I have had on a test bed for the last week. Its a great phone, with a good user interface, both physically and its web based one. At its lastest firmware it does have a few quirks, with regards to the way it handles usernames and passwords on the physical interface. These have

Hi list, Does chan_capi work with kernel 2.6? The Eicon Diva server card loads fine judging from /var/log/messages but Asterisk gives an error when trying to load the chan_capi module. I'm using chan_capi-0.3.5, zaptel-1.0.0, libpri-1.0.0 and asterisk-1.0.0 on a Fedora box with kernel 2.6.8-1.584. Zaptel and ilbpri work fine as does *. I have seen a msg that may be related and don't know

Hi all, i use asterisk 11.14.0 and I suspect that the voicemail application counts the time wrong. In my voicemail.conf: [general] minsecs=3 maxsilence=5 format=wav maxsecs=180 silencethreshold=140 [...cut..] In the asterisk-cli: [Jan 26 15:23:49] -- Executing [s at macro-voicemail:77]VoiceMail("SIP/XY-0005175a", "aNumber,su") in new stack [Jan 26 15:24:04] --

Greetings All, Regarding this archived post. http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html Did anyone ever find an solution to this? I've got a new box running 13.3.0 with the exact same issue. For those that don't read the link. I've got SIP Peers in realtime. All with a mailbox set. 98% of the time, These are loaded into asterisk without

Hi *, I want to integrate the Eicon Diva 4Bri Card to Asterisk. Eicon drivers and capi is installed. I use the latest dev version from eicon compiled and installed for my fedora 2 system. I found the chan_capi for asterisk from www.junghanns.net. Also loaded the patch and applied to the chan_capi source tree. I changed the Makefile to include the capi20.h from eicon:

Hi folks.. just a few weeks ago I wrote this to someone else: ------------------------ We have several 2900s in production as VoIP servers.. no lockups. On every server I go into the BIOS and: * Disable USB * Disabled uneeded things like Parallel, Serial * Put ETH0 on a seperate IRQ from the Digium card And everything's fine. Dell's do NOT have to share IRQs... go into your BIOS and

Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP function and assign it to a variable in an AGI script. I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's treating DUNDILOOKUP literally, rather than calling a funciton. I've tried this: EXEC "Set" "DIALPATH=${DUNDILOOKUP(2944093|180net)}" and also: SET VARIABLE

Hi, I have installed Asterisk 1.4.20 on Debian Etch. The server has no telephony card installed, but I have anyhow installed Zaptel (Zaptel-1.4.9) in order to be able to use MeetMe. The Zaptel modules load normally. I obtain the following prompts: kerplunk:/# /etc/init.d/zaptel start Loading zaptel framework: done. Waiting for zap to come online...OK Loading zaptel hardware modules: ztdummy.

Hi, in the Snom FAQ I found the following information: After staring up, the phone tries the URL given in the "Setting URL" of the phone. ... BTW this setting can also be set via DHCP. .... option tftp-server-name "http://192.168.0.9/snom200{mac}.htm" The documents used: FAQ-04-06-14-sf.pdf "Setting up DHCP for snom phones" FAQ-04-03-24-sf.pdf "How can I

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

On 26-10-16 15:03, Dan Jenkins wrote: > > > On Wed, Oct 26, 2016 at 1:46 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > > I keep getting the following error when trying to connect to the > Asterisk server using AMI : > > $socket = fsockopen("tls://11.22.33.44 >

Try different entry in this parameter. In Italy mobiles start with 3, while public services with 1 and normal user numbers with 0. Using pridialplan=none, every number different from 0 was resulting in termination code 1, normally used for number never seen on the network. Ciao Mauro

Does any one regularly reboot GS101? It sometimes lost registration with * and needs to be reboot. What is the best way to do it by cron? David Kwok

Hi All, I'm tring to do some DB operations before and after a call. I see the 'g' option in dial to continue in context if the destination hangs up, but what if the originator hangs up? Basically I do a DB get/put before the call is placed. After the call is completed I want to do another get/put; however the dial application dies when the originator hangs up. Any way to get around

Since upgrading from stable to latest cvs I can't place CAPI calls (AVM Fritz/chan_capi-0.3.1) Did I miss something that has to be changed in configfiles? Also tried to recompile chan_capi which run into an error. capi info shows me: Contr1: 2 B channels total, 2 B channels free. Any suggestions to these logfile snippets jo ---------------------------- * to ISDN -- Accepting

I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled ("1"). During startup * tells me that it connects to the db, so this should be fine. Nevertheless I don't see any users from the db when I run "sip show users" or "iax2 show users" although I configured some. It is also not possible to call them. Any hints?

Hello, I'm in the process of trying to get Festival to work with Asterisk. I followed the install process at http://www.voip-info.org/wiki-Asterisk+festival+installation. To get the Festival to compile I had to add the patch described in the comments. Once added, Festival and the Speech tools compiled without error. How ever, when ever I try to call the test extension, I get a busy