similar to: usage of manager events to create custom reports

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

Hi For whatever reason we've getting 2 or 3 CDR lines logged for each call, often in different formats: as1:~# grep test-89-1e2c /var/log/asterisk/cdr-csv/*.csv

Hello Asterisk Users, I guess the subject says the most of it; here goes some more detail: - Running Asterisk 1.2.14 - Objective: record all calls managed by a specific queue - Name those files ${TIMESTAMP}-${CALLERIDNUM}-${UNIQUEID} Facts: - If the UNIQUEID chan var is used in the MONITOR_FILENAME, before calling the Queue() application, the two legs of the call are not

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout |

Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the

Hi can you help me to build a asterisk manager command event to join a conference? i've seen that there is the event Event: MeetmeJoin Channel: <channel> Uniqueid: <uniqueid> Meetme: <meetme> Usernum: <usernum> Can you explain me how it works? Can i use it to join an existing conference? Thanks

Hi., do you think that sbr policy in queue strategy will be useful? Bye ---------- Forwarded message ---------- From: nik600 <nik600 at gmail.com> Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com> Hi to all isn't there any plan to add the Skills Based Routing strategy in

Hey All ~ 1, queue records on fairly unreliable. I would say about 40 - 60 percent of the queue calls are not being recorded and I'm not sure why. I don't seem to see any kind of pattern to the failure. I've added a sample of our queue config at the bottom. 2, cel_pgsql module seems to crash regularly. It seems every time I look at our asterisk server, the cel_pgsql module is

Hi to all the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? This is a piece of my sip.conf: [202] type=friend secret=202 host=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried

Hi, I guess I need some help with management interface. I would like to watch calls through the management interface, but I don't know how to identify which call an event belongs to or in other words how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/12125551111@pbx1 callerid: 12125551111 MaxRetries: 1

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

Hi to all i'm using PlayDTMF with AJAM, after the authentication, i make a request like this: host:8088/asterisk/mxml?action=PlayDTMF&Channel=SIP/200-sdadsadkioah&Digit=1 the result is: <ajax-response> <response type='object' id='unknown'><generic response='Success' message='DTMF successfully queued' /></response>

hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk is behind NAT - Asterisk ip is 10.10.10.2 - SIP user view Asterisk as 10.10.15.2 (Asterisk is behind NAT) When the CCM calls the SIP user the call works perfectly. The

Hi to all i'm using Asterisk 1.4 and need to announce something like 'The operator answering to you call is XXX' to the caller, is it possible to do that using an AGI script ? The syntax in Asterisk 1.4 is Queue(queuename[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the

Hi how can i retrieve the call unique id of asterisk in the dialplan? I have enabled the cdr logging on a postgres database. In the table cdr each record has a field that assumes an unique id (for example: 1141628669.51) Can i retrieve this from the dialplan? For example: exten => 203,1,Answer exten => 203,2,Set(CALLERID(name)=UNIQUE_ID - ${var_name_unique_id}) exten =>

Dear all i'm planning a transparent migration from a courier server that provides both IMAP and POP3 access to users to a remote dovecot server with both IMAP and POP3 access. I have to migrate about 2500 users for 250 GB of space. I'm using dovecot 2.2.5.4 on debian6 squeeze. To make a transparent migration i have to maintain old IMAP UIDs and POP3 UIDs, so i've read

Guys. Anybody knows why this is happening? Seems every time I make an internal call, the manager shows this and I don't get the callerid on my identapop but rather the calledid.. Event: Dial Privilege: call,all Source: SIP/intruder1-85f0 Destination: SIP/test-f037 CallerID: 201 CallerIDName: Anton Krall SrcUniqueID: 1117038116.7 DestUniqueID: 1117038116.8 Event: Newchannel Privilege:

thanks John yeah, your approach is much siple, i've tried it but i'm not able do detect DTMF tones. it seems that on calls that i receive DTMF tones are handled correctly, but on calls generated from Asterisk to the world when the called side sends some DTMF digits they are not detected: -- Executing [s at macro-myconnector:1] NoOp("SIP/pbx2-000004b2", "") in new