similar to: DialPlan for Call Limit, Call Duration, And Group Call

Hi Everyone, I don't see any parameter for limiting duration of a call in the .call file for Asterisk spool outgoing directory. I'd rather not use a MeetMe to drop the call in a conference room and to then limit the call duration as that complicates things unnecessarily. I am wondering if there is anything else I can do or if the "Channel" parameter take call duration like the

Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Using Asterisk 1.8.12.2 Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? -- Suich

Hello, I was wondering if there is a simple way to limit the number of simultaneous calls in an Asterisk PBX ? I've seen that we can make this easily per channel (like in SIP.CONF) : incominglimit=X, but I'm looking to limit the maximum calls all channels together. Another thing. Working with asterisk-perl, I need to get the call duration, currently I use

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, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

My question: How do I configure AAH via AMP to make a connection through our legacy PBX to the PSTN? Details: We're trying out Asterisk through Asterisk @ Home. Our legacy PBX has a modem type dial tone port that we hooked a Digium FXO to. Now I can dial from the XTEN client on my computer to any legacy PBX extension. If I connect a regular phone to the modem dial tone port, I can dial

Config: AAH 2.2 Digium TDM card connecting to 3 x Telus POTS lines Polycom 501 phones pretty basic setup, working mostly just fine... When I dial a number such as: 96045551212 Telus automation will sometimes come online and tell me that the number I have dialled cannot be completed as dialled. If I hang up the Polycom 501 and redial the EXACT same number, it will work the second time. I

Dear All, I' I have a problem in installing asterisk 1.4.0. how can i compile res_config_mysql.c in astersisk-addons dir. I've downloaded asterisk-addons-1.4.0 compiling and installing it. But i can't found shared object of res_config_mysql.so. My system is : Debian Linux 3.1 Kernel 2.6.8-11 asterisk-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 libmysqlclient using apt-get webserver :

Hi all, We use AAH to run our office telecoms registered with two Sipgate accounts. Today, Sipgate appeared to have had problems with their server with oneway audio on every call. In order to cause the Sipgate message service to pick up in stead of our AAH box, I simply unplugged the network cable. We now have problems where AAH does not seem to access the network. I plugged the network cable

Hello everybody. Recently I've been trying to limit the duration of some calls for a simple application I'm writing. Unfortunately all of the documented methods are failing and I'm not sure what else to try. Here are some samples of what I've done: 1) The AbsoluteTimeout application. - exten => 1,1,AbsoluteTimeout (30) 2) The new version of AbsoluteTimeout. - exten

Dear R helpers   I have following csv file which is an input   id       par_value    coupon_rate     frequency_coupon   tenure    ytm   1            1000             10                      1                     5          12   # Here frequency_coupon is coded s.t. 0 means Daily compounding, 1 means monthly compouding, 2 means Quarterly, 3 means Half yearly and 4 means only once. Thus in the

Hey all, is it safe to run the asterisk-update.sh script that comes with AAH to upgrade only the asterisk binaries? Doug has chimed in a few times saying 'upgrade' when I post problems, but Aah makes this really painful. I'm using AAH 2.0 & am fighting a number of 'bugs' that only seem to be manifesting in my installation. Can I safely upgrade just asterisk and not any of

First encounter with * Just downloaded & installed aah-2.7 Started up AMP, but i can not find any reference towards isdn. I presume there has to be some configuration done for my Eicon-Diva-pro. Does aah actually support isdn-bri? On the mail-archive i found some references, but these are rather old ( they speak about the coming release of aah-2.1) aah-handbook (version 1.6) doesn't

I'm still a learning but I purchased a dph-140s to test with AAH 1.5. I think this is a rebadged ACT P104SLD which others seem to have working with *. It seems to be configured and registered similarly to the softphones I've been using just fine, but it does not receive or send audio (it will send audio to vm), or perform the loopback test. It seems to signal and receive calls fine. The

Question 1: Is there a way to keep an authentication to a Samba share from lasting indefinitely? I am implementing a Linux file server for a network of machines all running Windows 98 SE. (Redhat 9, Samba 2.2.7a, share-level security.) Once a user provides a valid password and connects to a Samba share, Samba allows that client machine to access that share indefinitely. In our office each

Hi, Just set up AAH 1.1 using an HFC BRI line and 5 IP phones as per http://voip-info.org/tiki-index.php?page=ACT+P104+IP+Phone The dialplan was configured through AMP and has nothing fancy in it. As a first time user of not only Asterisk, but also a PBX, there are some operator teething problems. After much googling & searching of voip-info.org, I cannot find any answers to these

Does anyone here use either Gradewell or inWeb for service? They are both UK based. I'm trying to get a couple of inbound IAX2 based numbers from both of them to work with no luck at all. The one thing that sets these guys apart from the rest of companies offering inbound numbers is they tie the account to the IP of the asterisk server. Neither use register lines in iax.conf, there appears

Dear R users, I have noted a difference in the merge distances given by hclust using centroid method. For the following data: x<-c(1009.9,1012.5,1011.1,1011.8,1009.3,1010.6) and using Euclidean distance, hclust using centroid method gives the following results: > x.dist<-dist(x) > x.aah<-hclust(x.dist,method="centroid") > x.aah$merge [,1] [,2] [1,] -3 -6

Hi; I've got an AAH installation where a customer wants to install an active Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel 2.4.21.37. Support for Eicon active cards is built-in. I've debugged and run the A@H install-Eicondiva script but when I try to run divactrl load -c 1 -f ETSI -Debug I get a response : A: can't get card type for DIVA adapter