search for: mobillion

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

...ence suppression in X-lite ("Transmit Silence"=YES) the problem in X-lite was over. Does anybody have the same problem with X-web lite and does anybody have a solution for this problem. Or does anybody know an other embedded web based SIP client? Kind Regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: arjan.kroon@mobillion.nl internet: www.mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.di...

...Asterisk 1.0.0. At the web page http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial I found it is possible to use the parameters w or W. But when I place this parameter in my dail command it doesn't do anything. What am I doing wrong? Kind Regards, Arjan Kroon Mobillion B.V. email: <mailto:arjan.kroon@mobillion.nl> arjan.kroon@mobillion.nl internet: www.mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050524/06a9c181/attachment.htm

...ad command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede email: arjan.kroon@mobillion.nl internet: www.mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/71afee34/attachment....

...risk works again. If I look in the log files it look like that asterisk will 'hang or freeze', if two callers calls exactly at the same time the same perl function. Does anybody now if res_perl is multithreaded? (I use res_perl 3.0 and asterisk 1.0.0) Thanx, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: arjan.kroon@mobillion.nl internet: www.mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.di...

...lication from the commandprompt Hello, Is it possible to start an asterisk application from the command prompt? This application has to dial to a number. When the calling party picks up the phone, the asterisk application had to play certain voicefiles. Kind Regards, Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: arjan.kroon@mobillion.nl internet: www.mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.di...

...risk. Now I'm looking for documentation for the perl commands. Some perl commands I found on this URL: http://www.voip-info.org/wiki/view/Asterisk+PHP Does anybody got more documentation or where I can found some more documentation about perl commands Kind Regards. Arjan Kroon Mobillion B.V. Copernicuslaan 30 Postbus 554 / PO Box 554 6710 BN Ede tel: +31 (0)318-648920 fax: +31 (0)318-648839 mobile: +31 (0)6-55871460 email: arjan.kroon@mobillion.nl internet: www.mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.di...

...(es) are down? AGI(..) prints "Unable to locate host" and the dailplan jumps to extension h. I'd like to handle the return value and keeping the caller in the dailplan and not to the hangup extension. Any tips about how to handle a AGI(..) returns -1 condition? thx Arjan Kroon Mobillion BV -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090626/7589f3b5/attachment.htm

...still '655871460' Is there a way to reset the CALLERID(dnid) on one channel or automatically reset the complete cache on one channel if this channel is ended? Regards, Ami command: action: GetVar actionid: 129675971_656137# variable: CALLERID(dnid) channel: DAHDI/11-1 Arjan Kroon Mobillion BV

...D with asterisk version 1.6 The following bug was for version 1.4, but I cannot make up if this bug is still in version 1.6 http://forums.digium.com/viewtopic.php?t=7780 In version 1.8 it is possible to change the Connected Line ID, but this isn't the case in version 1.6 Regards, Arjan Kroon Mobillion BV

Hi, I can't read a environment variable in a asterisk dialplan. When logged in as user root on the system an 'echo $HOSTNAME' gives the hostame of the machine. Asterisk (1.4) is started from the same console. I try to read it like this: exten => s,n,NoOp(host=${ENV(HOSTNAME)}) Does anyone know what i am missing? Ipv een saaie e-mail een leuk videobericht? Ga naar

...guration file, just say which configuration files (sip.conf, etc) First I tried to a simpler situation, see below Caller --> asterisk00 (inbound/outbound server) --> SIP client (X-lite) This situation worked perfect. Thanks in advance Arjan Kroon email: <mailto:arjan.kroon@mobillion.nl> arjan.kroon@mobillion.nl -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050519/fa29873a/attachment.htm

Hi, I'm using videocalling on asterisk 1.4.10. When I setup the videocall with exten = n,1,h324m_gw(s at video2webanswer), I loose the variable DNID (${CALLERID(dnid)}) Before the videocall is set up, this variable is filled and after this videocall this variable is empty. Also all local variables are empty. If al look at the A-number (${CALLERID(num)} this variable is not empty

Hi, In my application I jump to different extensions For example: [begin] exten => s,1,Goto(starts,s,1) [start] exten => s,1,Play(welkom) ..... exten => h,1,Goto(end,s,1) [end] exten => s,1,Macro(end_call) exten => s,n, Hangup When I look at my CDR record I see three different CDR's in my record. Is there a way to use one CDR on every call, and not

...kill all rawplayer sessions Does anybody have the same problem with this problem. A way is to upgrade asterisk, but this is not now the solution for us. The code for the rawplayer is: /usr/bin/rawplayer #!/bin/sh for name in $@; do cat $name ; done Regards, Arjan Kroon Mobillion BV -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100216/21820a83/attachment.htm

...requested but no musiconhold loaded. [Dec 24 14:34:03] NOTICE[9030]: channel.c:4006 __ast_read: Dropping incompatible voice frame on SIP/arjankroon-00000000 of format gsm since our native format has changed to 0x4 (ulaw) I'm using asterisk 1.8 Can anybody help me? Kind regards, Arjan Kroon Mobillion BV

Hey there, I'm trying to convert some call recordings from asterisk we have in .gsm format to something I can pipe through ffmpeg - wav would be good, mp3 would be amazing! I've been trying playing with sox but I don't seem to be getting too far with 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample as ffmpeg borks at it: tim at freee-meee:~/dmc/call

hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------

Hello, I'm a newbie in connection several asterisk servers with each others. I've got the following situation. I've got 9 asterisk servers (asterisk00 till asterisk08). When I call to asterisk08 then I want to redirect an application which runs on asterisk00. But how can I redirect in an application on asterisk08 to an application on asterisk00? Or isn't this possible?