similar to: Queues with announce

Hello *, I have a strange problem with the MAPI proxy AstManProxy: sometimes it happens that I send a request and I receive a response to ANOTHER request that it got in the frame time between my request and my response. Did anyone else notice this behaviour? How can this be solved? I've been reading the source code, but I didn't find a solution. Thanks in advance, -- Dr. Andrea

Hello everybody, I'd like to have more detailed records for calls related to queues. For instance, if A enters in queue X, waits for Y secs and then talks to peer Z for T seconds, I'd like to have two entries in my CDR: - src: A, dst: X, duration: Y, state: ANSWERED - src: A, dst: Z, duration: T, state: ANSWERED This independently from how many peers the Queue app calls without success

Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute

Hello everybody, I'm trying to understand how can I set the MoH class for parked calls. I set the "incoming" class for calls, and it works correctly. When I park the call, the music on hold is ok, but when I close the communication on the parking side, the parked call gets the default music on hold class. Can someone explain me what's going on? Thanks in advance, -- Dott.

Hello everybody, I'm aware that I can try to balance gain levels with PSTN cards using the ztmonitor tool, as described in http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html (Adjusting the rxgain/txgain Settings). Is there a similar tool for mISDN? If not, what is your approach to gain setting in mISDN? Thanks in advance, -- Dr. Andrea Spadaccini

Hi folks, I'm trying to install asterisk with radius cdr support. I got freeradius up and running, so following radius instructions inside asterisk source package, I've installed radiusclient-ng and relative headers. But when I start configure(asterisk 1.4.18.1) I got: checking for rc_read_config in -lradiusclient-ng... no If I type: ./configure --with-radius=/usr/share/radiusclient-ng the

Hi guys, Does someone use a mysql database for accessing data and in the same time for storing cdr ? if that is the case, which module is used ? thanks -- Cyril SCETBON

Hi folks, I used the ast2sql.pl script (found on www.voip-info.org) to put into the database a simple sip.conf. Among other entries, you could find: [general] context=sip-in ;incoming sip calls Well, the script put the comment into the database entry, and asterisk started complaining about a 'sip-in ;incoming sip calls' context not found in extensions.conf. IMHO the comments should be

I've asterisk stop (presumably segfaulting) a couple of times, and I was just beginning to look at how to keep it running - what have others done? I was thinking of wrapping a script around asterisk like this: while 1 do asterisk -f done /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Ahoy I'm trying to setup Asterisk on debian etch (with the debian packages) with a "Fritz!Card PCI" ISDN card and chan_capi. Everything seems to be configured the right way (excerpts below), Asterisk seems to see the ISDN-card but if i try to place a test-call from the outside i don't see anything on the asterisk-console (set debug

hi there :) i get this error in the asterisk CLI: Sep 21 20:31:49 ERROR[3774]: cdr_addon_mysql.c:159 mysql_log: cdr_mysql: Unknown connection error: (2006) MySQL server has gone away when I run 'cdr mysql status' i get: Connected to voiceone at localhost, port 3306 using table cdr for 18 minutes, 24 seconds. Wrote 8 records since last restart and 1 records since last reconnect. I

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

#1 is the most important but I'm already debugging that. I don't want to interrupt your holiday but if you could look at #3 that would be great. > As to the issue #3 - do we need to keep compatibility with the old mips-mti-linux-gnu toolchain layout? It would be nice to support both the old and new layouts but I doubt that's possible with the current framework (see below). If

Hi folks, I'm really confused, so please help me, or at least give me some pointers to clarify this issue. Can I mix "Static" and "Real" realtime? Is there a way to easily switch from one to another, say, for sip.conf? Which are the major benefits of "Real" realtime? Please help me! Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute

Hi Daniel, I am on a vacation now till the Aug 3 but I can take a look at these problems. What is the most important? As to the issue #3 - do we need to keep compatibility with the old mips-mti-linux-gnu toolchain layout? Simon On Wed, Jul 29, 2015 at 1:08 PM, Daniel Sanders <Daniel.Sanders at imgtec.com> wrote: > The issues are: > > 1. Almabench has some significant

Hello everyone, I'm Andrea, and I've started working with Asterisk a couple of weeks ago, so I'm still a newbie. :) I was reading about Asterisk Realtime, and I was wondering if I can mix Static realtime and "Real" realtime configuration. For instance: can I have a "Static Realtime" extensions.conf and use "Real Realtime" sippeers and sipusers? Moreover,

Hello, I've enabled BT-200's SYSLOG logging, and I get some message whose meaning is obscure to me. In particular, in a day I got the "Deletion of invalid timer" message almost ten times from one phone, which has some call problems. Can someone point me to a resource on BT200 error codes? Thanks, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l.

Hello guys, as you probably have already understood, I'm trying to make asterisk realtime work. Well, now it's working, but I'm not fully understanding the "metrics". In voip-info.org I found that they are a sort of "position" inside a context (var_metric) or the index of the context (cat_metric). Am I right? Where can I obtain more info about these

Hello everybody, I want to use the TIMEOUT() function, but in the CLI the "show functions" command only shows 7 custom functions: QUEUEAGENTCOUNT SORT CUT CHECKSIPDOMAIN SIPCHANINFO SIPPEER SIPHEADER In addition, sometimes I get the debug message "function LANGUAGE not registered". How can I install those functions? I'm using Asterisk 1.2.10. Thanks in advance, --

Hello everybody, I'd like to hear some success stories about the use of Asterisk Realtime in medium-large contexts, like > 50 extensions. Don't you think that in those contexts the system could be overloaded from the excessive number of queries to the DB? So.. is anybody using ARA in those kind of deployments? Thanks in advance, -- Andrea Spadaccini Multimedia Technologies