similar to: Voicemail Realtime issue "Failed to obtain database object for"

Hello everyone, I would be extremely glad if someone could help me with the following issue: cat /etc/asterisk/func_odbc.conf [call_user] prefix=GET dsn=asterisk_odbc_sip readsql=SELECT name FROM asterisk_sippeers WHERE name = '%477' Asterisk CLI show the following issue: -- Executing [205 at phones_wildcard:1] NoOp("SIP/Y_MD_vlungu_477-00000008",

Hello everyone, I would appreciate if someone could help me with the following issue: http://pastebin.com/bTskMLVw My res_odbc.conf file look as follows: http://pastebin.com/bhReQkXQ -- Mit freundlichen Gr??en / Best regards Vadim Lungu *System Engineer* Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lungu at yopeso.com <mailto:vadim.lungu at yopeso.com> Web

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:

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

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

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, 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

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

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single video source of the conf. The video mode stays follow_talker. I

Hi All, I need a softphone (PC/Mobile) which does auto redial in any case (noanswer, answer, busy, congestion etc) after a given time interval. So if the time interval was 5 secs, it would dial last number dialled after every hangup (or every failure to dial). Does anyone know such feature in a softphone? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi

Hi Everyone, How can I create a TLS based sip trunk between two asterisk servers. I have been trying to do it but i dont know how to defined the client certificate on the asterisk server. Has anyone tried this? -- Best Ragards Rizwan H Qureshi V: +971 (0) 528272154 linkedin.com/in/rhqureshi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have a GSM to VoIP gateway (specifically yeaster TG400) which I am trying to use for kind of a call intercept between two GSM users. Call comes through one SIM and goes out through another Sim with our Asterisk in between to log the call. This works fine but we need the original callerid to pass-through through the outgoing SIM. I have tried every possible configuration on Asterisk that

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>

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

Hi, I use tinc-vpn to create private mesh vpn networks between office and house in different locations and it works really well. But somehow I'm considering if there were any possibility to add radius support for single node traffic management. I does think it will be really difficult to implement such functions as a feature of a mesh network,because all of the traffic was initiated end to

Hi all, A week or so down the list, i read that not many people were using realtime on an Asterisk18, so i had this afternoon a go at it... [sorry for the inconveneant line-wraps] First i did: mysql> create database asterisk; mysql> grant all on asterisk.* to 'voipadmin'@'localhost' identified by next i used the info from the wiki: CREATE TABLE `sip_devices` ( `id`