similar to: Call was hangup when LIMIT_WARNING_FILE was playing

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: --

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]

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

Hello all, We have a customer who upgraded from Asterisk 1.6 to 1.8, and pickup groups which previously worked fine have stopped working. Can anyone advise if there has been a change in how pickups work? Here is an example where 1000101 is trying to pick up a call to 1000103: <SIP/product-local-00000005>AGI Rx << EXEC Dial "Local/1000103 at product-pickup

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 fellows.. I have 2 asterisk servers in which the following line exten => _09049.,111,SetVar(LIMIT_PLAYAUDIO_CALLER=YES) exten => _09049.,112,SetVar(LIMIT_WARNING_FILE=beep) exten => _09049.,113,Dial(${TYPE}${DESTINO}|30|L(30000:10000)) works OK on my Asterisk 1.2.9, it plays the beep 10 seconds before the end of the call. doesn't work on my Asterisk 1.2.13, it hungs 10

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

Last question for today, I promise! The problem: In order to disconnect calls after x minutes, I need to do this: [setup] exten => setup,1,Answer() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached) same => n,Dial(Local/s at root/n,3,L(3540000:60000)) same => n,Hangup() [root] exten

Hello, I want to limit calls per sip account user. How may I realize this setting? For example I want to limit to 10 min all possible calls from an account or to limit external calls to 10 min and local call remain unlimited. Thank you for support guys. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

Asterisk 14.1 Here's a bit of test dialplan, which works as expected and simulates exactly what I'm doing at the top of my large dialplan... [dial-pre-test] exten => s,1,NoOp() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=time_limit_reached) same => n,Dial(Local/s at dial-test,3,L(3540000:60000)) same => n,Hangup() [dial-test]

Good Day List Users, Is there any way to play an audiofile or at least a beep into an established call, I want to do this event each 3 minutes in the call, for now I have a shell to get the call time and evaluate the 3 minutes.....do you know any way to play that sound? I tried app_inject, it works really nice in asterisk 1.4.X releases; but my PBX runs 1.2.14 and It can?t be upgraded (policy

Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I'm trying to find out a way to whisper the time remaining for a prepaid application on a established channel. Unfortunately I think there is a lack of PlayBack/Background commands which can be applied on a working channel as well as a lack of spy/whispering commands available via Asterisk Manager. Does anyone know how to implement this? Thanks a lot. Regards, Victor

Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped

Hi all, I try to make a call from my Openser(SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip

Thank you - that makes sense. I've seen something about swapping and optimizing channels on the console, but I didn't realise "optimize" meant "not do what you wanted". OK, so here's why I'm dialling anything at all: The first dial is because I MUST limit the incoming call to less than 60 minutes. The second dial, which carries the gH option, is because I