similar to: Change to pickups in Asterisk 1.8 - not working on local channels?

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Hi, we're using the limit option like this: Dial ....L(60000:30000) [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] -- Limit Data for this call: [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] > timelimit = 60000 [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] > play_warning = 30000 [Jun 10 16:14:41] VERBOSE[12196]

Hi, I?m playing with the CVS head time limiting at Dial application, it just works fine but the only problem is that the caller isn?t hearing the warning message. I?m using a Cisco 7960 as the caller and a Polycom 500 as the callee. The audio is passing through Asterisk: -- Executing Dial("SIP/8992-9712", "SIP/8988|20|L(10000:2000)") in new stack -- Limit Data: --

Dear All, I tried to use 'L' option on my dial command. I set the x to 65000(65 seconds), y to 60000(60 seconds), z to 30000(30 seconds). The max calltime should be 65 seconds, and it will play "beep.gsm" at 60 seconds left. And repeat the beep at 30 seconds left. But the call will be hangup by system at 60 seconds left. In another word, when it plays warning file, the call

Here i am doing a dial command from a php agi... EXEC DIAL H323/123456789@xx.xx.xx.xx:1720|40|HL(585000:61000:30000) But asterisk is not disconnecting the connection after 585 secs... the result is ... answered time is 1926n but thing is time out is working some time and some time not.... LOG: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: callcard.php: string(111) "app_callingcard:

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

Hi, i want to parse a XML-File. I made some Tutorial but with my special Format it don't work. An Example of my format: <?xml version="1.0" encoding="ISO-8859-1"?> <mzML xmlns="http://psi.hupo.org/ms/mzml" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="http://psi.hupo.org/ms/mzml

Hi all, how can I pickup a call from a queue? Which context parameter do I have to specify? The context that calls the queues application is ext-queues. This is what I tried so long (777 is the extension of the queue I want to pickup from): exten => _**ZXX,1,Noop(Attempt to Pickup ${EXTEN:2} by ${CALLERID(num)}) exten => _**ZXX,n,Pickup(${EXTEN:2}@from-did-direct) exten =>

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Hi all, I have a GXP2000 with BLF configured. I follow the configuration guide to enable the pickup cmd as follow and include it under corresponding content. [BLF_group_pickup] exten => _**1XX,1,Pickup(${EXTEN:2}) exten => _**1XX,n,Hangup The I press the single key to pickup the call to extension 100 when there is a call to it. From CLI, I can see it issue **100 to asterisk but failed

Hi, I'm using an Aastra 57i together with Asterisk 1.4.13. The 57i is configured for call pickup as recommended by Aastra. When the LED flashes, I press the corresponding button and the display tells me "Call not possible". In the CLI is see the follwoing output. Why isn't the call transfered to user2 and what does "No target channel found for 31 mean"? User2 can

Hello, I searched this list archives and couldn't find any practical way to disable newly introduced dialog-info based call pickups (see CHANGES file). Suggestions ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100426/cc9dc011/attachment.htm

Hi there, i have a Problem with the Pickup command. Versions: asterisk 1.4.1 on gentoo my extensions.conf [only the interesting part]: [incoming_1] exten => 123,1,Ringing exten => 123,2,Dial(SIP/xxxx,20,r) exten => 123,3,wait(90) exten => 123,4,hangup [incoming_2] exten => 456,1,pickup(123@incoming_1) both are sip-accounts and have pickupgroup=1 in the sip.conf so my idea is,

Hello, We have a situation where a call comes in, users are notified via an external process (curl request to web service), and we can't answer the call until a callee can call in and pickup the call. How can we implement this functionality? We tried using : [caller-inbound-leg] ; code to send the CALL_UUID information to users. exten =>

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

Hello, all. I've got a problem where we set up call pickup for a customer. If the Bob's extension rings and Bob is in Jim's office, Bob can press the button on his Snom 320 that says "Bob" and pick up his line. It works great for calls coming in from the outside but does not work for internal calls. Internal calls generate a app_directed_pickup.c:204 pickup_exec: No

Hi all, How can i write a puppet code such that..it adds a line of content that i want..at a desired place in a file.. if [ "x$JAVA_OPTS" = "x" ]; then JAVA_OPTS="-Xms64m -Xmx512m -XX:MaxPermSize=256m -Djava.net.preferIPv4Stack=true -Dorg.jboss.resolver.warning=true -Dsun.rmi.dgc.client.gcInterval=3600000 -Dsun.rmi.dgc.server.gcInterval=3600000

On 4/7/20 11:54 AM, Gary Stainburn wrote: > I have fail2ban on my mail server monitoring Dovecot and Exim. > > I have noticed that it has stopped banning IP's. I have seen in /var/log/fail2ban.log: > > 2020-04-07 09:42:05,875 fail2ban.filter [16138]: INFO [dovecot] Found 77.40.61.224 - 2020-04-07 09:42:05 > 2020-04-07 09:42:06,408 fail2ban.actions [16138]:

A quick search on Google gives: http://www.unixfu .ch/how-do-i-update-the-root-hints-data-file-for-bind-named-server/ Best regards, Marcel de Reuver 2017-06-06 11:36 GMT+02:00 Torsten Kurbad via samba <samba at lists.samba.org>: > Hi, > > we are running a Samba AD on UCS 4.2, which comes with Samba 4.6.1. > > The DNS server (192.168.0.200) is operated by bind with the samba