similar to: R: queue behaviour

Hi everybody, I'm facing a strange problem after upgrading Asterisk from 1.0.9 to 1.2.4. Sometimes, when receiving an incoming call from pstn, although my sip phones ring correctly (I've got both softphones and hardware phones), noone can pick up the call. Asterisk CLI shows me that the phones are ringing, then nothing happens, so there's no problem _after_ someone picked up, simply

I have the following pair of macros defined to handle outbound calls from *. Rather than specifying full dialstrings in the main body of extensions.conf, outbound dial commands are made using a macro call as follows: Macro (outbound,number_to_dial,callerid_to_present,gateway1,gateway2,gateway3,gate way4) The final gateway defined is nearly always a fallback to PSTN if none of the IAX or SIP

Hi everybody. Just noticed that when calling a mobile phone, Asterisk doesn't bridge the voice message by telco if mobile is unreachable, but keeps on ringing till it receives a hangup signal. I think this is due to the fact that the message is played without the call has been answered, but I'm wondering if there's some way to let Asterisk realize it. All I see in the CLI is the line

Hi folks, need some help on queue behaviour. What I'm trying to do is accepting a call from pstn, put it into a queue, while callee is waiting contact some numbers till one responds, then bridge the two calls. What I can't manage is jump to next dialplan command soon after callee enters the queue in order to call other numbers. I also tried AGI and Asterisk Manager, with the same

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of

I apologize, but I can not find a complete list of directives in dovecot.conf possible. where can I find? thanks excuse my for my orrible english -- Caselle da 1GB, trasmetti allegati fino a 3GB e in piu' IMAP, POP3 e SMTP autenticato? GRATIS solo con Email.it http://www.email.it/f Sponsor: Apri Conto Corrente Arancio entro il 28 febbraio 2011 e ricevi 100 euro da spendere su Media

Hi Charles, attached there is a patch for a new version of riello drivers: Changes: - small changes in riello_usb.c and riello_ser that solved problem with UPS that not support realtime autonomy calculation (and also internal temperature); - the second problem I think is related to the first because the shutdown command that I sent to UPS by NUT monitor works without problems; Cheers, Elio

Thanks, chdir dir /data error seems go away, but at 12:00 o'clock the following error appear: Feb 20 11:45:03 brulx01 vsftpd: Mon Feb 20 11:45:03 2006 [pid 12273] [job260] OK DOWNLOAD: Client "10.90.1.1", "/data/mde/DATA/.././ACTUAL/mdeact260.msg", 12481 bytes, 174.76Kbyte/sec Feb 20 11:56:10 brulx01 smbd[12410]: [2006/02/20 11:56:10, 0]

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

Hi to all, i'm wondering to realize a dynamic macro that can take the number of extensions to RING,the ring type and all the parameter in a dynamic way. I have done this code to test it: macro pbx-ring-group-ael(pbx_id,num_int,ring_type,timeout,ext_string) { //; pbx_id = Id of PBX in the DB //; num_int = Quantity of extensions to ring //; ring_type = Kind of RING (C=contemporaneous

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

Dialing out If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. If you want your application to initiate a call out without being started through the dialplan: * Asterisk auto-dial

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output says: AGI Rx << GET VARIABLE DIALSTRING AGI Tx >> 200

hello! has this made it into 1.2.3 already: http://bugs.digium.com/view.php?id=6128 ? i'm trying to set a variable that should be used as a dialstring in the dial-command, including parameters seperated with the respective delimiter, e.g. like: exten => 907,1,Set(DESTINATION1=Zap/G1/4989123456789|10|gh) exten => 907,n,Set(DIALSTRING=${DESTINATION1}) exten =>

Because the VoiceBlue is only 4 channels and I am supporting 100 cell users, I needed a way to overflow calls to the PRI of all 4 channels are full. Unfortunately, there seems to be no built-in mechanism to determine if the gateway is full, so this script parses the output of "asterisk -rx sip show channels XXX.XXX.XXX.XXX" to determine the number of channels currently in use. Hope this

Hi I'm using a macro to dial in a AEL dialplan. The problem is the macro do not set the field CDR(dst), showing only ~~s~~. I tried various configurations, but without solutions. This is the macro: macro dial-out(destno,dialstring,route_descr,interno) { __TRANSFER_CONTEXT=ipbx; if(${interno} = 1) { Set(__PICKUPMARK=${destno}); if(${ODBC_verify_user(${CALLERID(num)})} > 0) { t = tT; }

On 22 febbraio 2014 17:28:16 CET, Roberto Resoli <roberto at resolutions.it> wrote: >Hello, > >after some days running without problems, my riello idialog 800 ups >get disconnected, and usbfs started flooding syslog with messages like: > >usb 1-1.2: usbfs: process 57342 (riello_usb) did not claim interface 0 >before use > >Version is 2.7.1-1 debian packages, which

ah... ok, now i understand.   So you have a small difference between these to servers. My first guess would be something with vfs_objects.  Wrong orders might hide the security tab in windows.   Your setting up windows ACL's. Remove all posix related setttings from smb.conf ( im asking Rowland for support here, because om on holiday in 20 min  )  :-P :-)) ...   All you need for a windows

I've got a macro that tries to find the first available SIP trunk to send outgoing calls on. It tracks the usage of the lines (since each trunk has a call-limit of 2) by using GROUP(). My problem is that once a call switched to ANSWER state, ``group show channels`` stops listing it and then my Macro starts screwing up because it's sending calls to a line that sometimes is full even