search for: dialopts

...out,Agent/${AgentNum}); } } context outboundQueue2 { .... } // outbound call logging for queuemetrics: macro qmoutqdial( clid, dialstring, queue, agent ) { start_dial_time = ${EPOCH}; QueueLog(${queue},${UNIQUEID},${agent},CALLOUTBOUND,-|${clid}); Set(dialopts=gWKU(queuelog_connect_event^${queue}^${UNIQUEID}^${agent}^${start_dial_time})); Gosub(pstnRouting,${clid},1); end_dial_time = ${EPOCH}; verb = COMPLETECALLER; &queuelog_hangup_event(${queue},${UNIQUEID},${agent},${start_dial_time}); return; } // central ca...

Hi, Iam dialing from NT ptp to SIP provider. Sometimes Asterisk is doing music on hold but there are no options like t or T in the dial command. As an result the channel got lost and an Hangup occurs. Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card. Any solution for this? Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered Zap/8-1 Oct 22 11:20:23 VERBOSE[29983]

How to implement some of the function into asterisk like *67 "call number blocking" or *57 "call trace" ? I'm connecting to sipura SPA3K outside line by dialing 9+number. Is there a way to get outside dial tone in SPA3K PSTN-Line by dialing "9"? How to program the extension? -- #Joseph

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??

Hi, I've recently discovered Dial examples such as "Dial(DAHDI/g4d/${EXTEN})" but I wonder where I can get an uptodate doc. Is there any CLI option such as "core show channel dialoption" that would explain what "g4d" exactly means ? "core show application dial" doesn't explain much about those channel-dependant Dial options. Regards --------------

Hi All, I am trying to call my own service through Asterisk and the DTMF is not recognized . I also noticed the following issue, the phone rings for about 8-9 times before the line is picked up but when it is picked up it seems that our system has picked up the call much earlier, I could just not hear anything except the ring. that means other system picked UP a call and my SIP phone still

...parties.agi: dbset CALLTRACE/38532 to 8688 > dialparties.agi: extnum 48532 has: cw: 1; hascfb: 0 [] hascfu: 0 [] -- dialparties.agi: dbset CALLTRACE/48532 to 8688 -- dialparties.agi: Filtered ARG3: 8532-38532-48532 > dialparties.agi: NODEST: 8532 adding M(auto-blkvm) to dialopts: trwTWM(auto-blkvm) > dialparties.agi: NODEST: 8532 blkvm enabled macro already in dialopts: trwTWM(auto-blkvm) -- <Local/28688 at from-internal-d4cc;1>AGI Script dialparties.agi completed, returning 0 -- Executing [s at macro-dial:7] Dial("Local/28688 at from-internal-...

Hi list, We're trying to set up a phone number that customers can call to get a hold of anyone of a group of sysadmins (and not their voice mails!). We found the findme example ([1]) that makes the callees press 1 to accept the call. It almost works, but it doesn't work correctly when one of the callees, the sysadmins, hangs up after accepting the call. We're using this

Hi gang, Since I'm getting no joy from device_Status or SIPPEER in 1.4.26-rc1, I thought I would do an AGI to read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet;

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

...G[15840] db.c: Unable to find key '206' in family 'CFU' Dec 21 13:43:48 VERBOSE[15840] logger.c: > dialparties.agi: extnum 206 has: cw: 1; hascfb: 0 [] hascfu: 0 [] Dec 21 13:43:48 VERBOSE[15840] logger.c: > dialparties.agi: NODEST: 600 adding M(auto-blkvm) to dialopts: M(auto-blkvm) Dec 21 13:43:48 VERBOSE[15840] logger.c: > dialparties.agi: NODEST: 600 blkvm enabled macro already in dialopts: M(auto-blkvm) Dec 21 13:43:48 DEBUG[15847] manager.c: Manager received command 'Logoff' Dec 21 13:43:48 VERBOSE[15847] logger.c: == Manager 'admin...

Hi all, Recently, I have upgraded the asterisk as following. asterisk-1.4.13 asterisk-addon-1.4.4 libpri-1.4.1 zaptel-1.4.5.1 Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh After upgrade, the server get segfault randomly and asterisk crash and restart itself. I got 2 core dumps of the segfault. Based on the core dump, we can't figure out the root cause to

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>