search for: ragarding

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Hi all, I am syncing call files on my secondary asterisk server but without permission to read for asterisk. So they should be executed when I grant the right permissions (thats when my primary asterisk server crashes or shutsdown somehow). But asterisk only tries to read the file at the time of placing the file. So when i grant right permissions nothing happens. Is there any workaround to this

Hi All, I have to forward incoming call on PRI back out to PRI but I need the original Callerid to passthrough. Is it possible with DAHDI PRI cards without involving the service provider? Thanks -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I am building an asterisk PBX with some stuff to make a usable VOIP / PSTN Gateway. I use the following devices: SIP Phones from GrandStream for VOIP side OpenLine4 from voicetronix for PSTN Side I am building things step by step with some priorities. I have now a working system able to place and receive calls from/to pstn. Before attempting to bring other functions (like voice

Hi all, Is there any way of originating calls in future without using call files? We have 2 servers (1 active at a time). If we use call files with modification date in future, on the 1st server and it dies and, we activate the second server but we lose the call files. I could have a cronjob on both servers and create callfiles reading execution time from database, but this involves some other

Hello list, i have one question related to call forwarding. i have 2 number for the inbound and i want to configure asterisk like that. When the customer call the first number 0522XXXXXX the call will be forwarding automatically to anther number 0520xxxxxx Does anybody have a solution to this problem. Thanks and Regards. -------------- next part -------------- An HTML attachment was

Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here <http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemail> and here <http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage>. I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for

Thanks AJ and David, We were actually using GSM gateways by setting busy forward number on the SIMs and just giving busy signal on every incoming call, telco took care of the forwarding and the line was free within seconds. Now we need to scale up the setup but GSM gateways a very very expensive if we want to scale upto a 1000 DIDs, which means thousand SIMs and a gateway/gateways big enough.

Hi all, The problem I have been experiencing since last month is that some of my customers are getting calls with "Asterisk <Unknown>" caller id. Most of them in the middle of the night. And my asterisk server has no record of these calls. The customers were getting irritated as you can imagine. I guessed the only way to receive incoming calls by by-passing the registration server

hello i am a new user for this group. i am working at a ISP. here i want to made a bridge firewall i am using fedora core 3. i want to block a serirs of ip address 192.16.18.0/255.255.255.0 and want to give the accesss only 172.16.18.0/255.255.255.0. but iptables not be able to block ip;s its passes all the ip series. i made my machine as bridge. i think my bridge passes all the

Hey Don, How are you? I may be heading your way in the next month or so. Have to meet with a guy in Eden Prairie, and stop off at my brother/sisterm-in-law's as well. Got a question for you - with TBCT, who pays for the call once it is transferred? Still me as the owner of the trunk? Lets say I take a call that was dialled locally (caller believes this is "free"), and I do a

sorry for this, but I want know: flac commandline syntax. support ordinary win. com.line ? other words: if I tape at encoding options --cuesheet=*_cdimage.cue, flacenc look it as '*_cdimage.cue' or as ''any'_cdimage.cue'? best ragards... "Opossum" <darckopossum@yandex.ru>

The config file indicates that if mail_location is empty, dovecot will try to find the mailboxes automatically (and by implication determine if the user is using mbox or Maildir format). But if the mbox inbox is non-standard (I have procmail delivering to ~/.mail) then one needs to specify mail_location = mbox:~/mail:INBOX=~/.mail and presumably dovecot will no longer try to determine what

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using

Hi all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications

Hi all, Do some one experiencing running jabber applications (jabberstatus...) in asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I got such result. IBM*CLI> help jabber No such command 'jabber'. IBM*CLI> help jabberstatus No such command 'jabberstatus'. Any one can help me on this, or may be I miss out somethings that cause jabber applications

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have a problem with Asterisk 1.4.x and the call manager. When I originate a call by the call manager or by a dot-call file only the calling party can hear the called party, not vice versa. When I dial the same number directly from the SIP phone (Cisco 7960) everything is OK. The same configuration worked with Asterisk 1.2 last week before

Hi there everyone, I am a bit confused these days due to some problem I am having. Its not a technical problem. Asterisk is working fine. Most of the users are happy, but some handful of users are getting calls in the middle of the night even though they have enabled "Anonymous Call Rejection (blocks calls with no caller id on asterisk server)" and TIMED DO NOT DISTURB which also blocks

I am using asterisk 1.8.3. I am getting this error: [Mar 15 09:49:12] NOTICE[1049]: chan_sip.c:21358 handle_request_invite: Call from 'mndemo_to_vizioconfrm104' to extension '1104' rejected because extension not found in context 'smvoice-mediaport'. "dialplan show" gives me that the context is present: [ Context 'smvoice-mediaport' created by

Hello!! I am a student from The Madurai Kamarajar University,doing a project in BioInformatics. I am now downloading the PDB FTP archive using the RSYNC,and am successful.I wanna know if rsync provides any script for the weekly updation of the PDB archive that takes place every wednesday 1.00 PM pacific time.I have at present the GETPDBUPDATE.PL script with me.but I want to know if rsync has got