search for: groupcount

hi, i have strange problem with call-limit/groupcount limiting. i set up limit of 2 calls. i'm using both methods but a for few times i have problem with successfull fraud with more calls than 2 asterisk is 1.8.22 someone with the same problem? any ideas how to solve or debug this problem? -- --------------------------------------- Marek ===...

...om 1.4.32 to 1.6.0.28 (as i think it should be the most stable now). Reading the UPGRADE-1.6.txt file i've noticed that: * SIP: The "call-limit" option is marked as deprecated. It still works in this version of Asterisk, but will be removed in the following version. Please use the groupcount functions in the dialplan to enforce call limits. The "limitonpeer" configuration option is now renamed to "counteronpeer". As i've experienced some problem with 1.4 release about call-limit, i'd like to test this new counteronpeer functionality, but how to handle t...

hi, i'm using groupcount to limit max calls in pbx i want show/graph made calls usage it's possible make this from cli/ami? something like asterisk>group show usage name channels group1 3 group2 5 google doesnt help thanks --------------------------------------- Marek Cervenka ================================...

Hi All, Can someone please tell me how to limit incoming calls to SIP channels using the SetGroup & Checkgroup command. I don't want any call waiting on SIP channels and you are somehow meant to be able to do it with these commands. Many Thanks Daniel Niasoff

Without call-limit defined, when a sip extension calls another sip extension then "show hints" shows that both are InUse (as expected). When one of them hangs up, both hints status become "Idle" (as expected). With call-limit=1 for each SIP extension: the caller is always Idle while the callee is InUse. Is this behavior normal? Doesn't sound right because if, during the

I am checking on the SetGroup / CheckGroup commands, but I have some troubles to undestand the examples. SetGroup(moh) can be moh anything as I like? Usually moh stands for "music on hold" CheckGroup(1) checks if somebody in in group "moh". Does it mean I can only have one SetGroup(xxx) ?? When I look at example 2 than I see two SetGroup commands and one CheckGroup

Is there a way to limit simultaneous calls. I like to limit simultaneous outgoing calls as more than few simulataneous calls are charged by my voip providers. However, I do not want to have any such restriction for internal calls. Thanks Jim

I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I

Hi all, I am just scratching my head trying to work out a way to use SetGroup to check busy status on a sip to sip call. The complication is that one call can't be in two groups so I have got no way of setting busy status on both the calling and called party. Has anyone got a way around this. Thanks Daniel -------------- next part -------------- An HTML attachment was

I have a question about the Asterisk Application ICES. I've got Asterisk setup to accept a phone call and call the ICES app which sends it to an Icecast server. exten => 1,1,SetGroup('stream') exten => 1,2,GetGroupCount() exten => 1,3,Ices('contrib/${GROUPCOUNT}-ices.xml') exten => 1,4,Hangup Everything works fine. Unless I have more than 24 phone calls being converted at the same time. When I try to bring up the 25 phone call the call comes up, Asterisk answers, and begins encoding the call. Eve...

Hello guys, I've one SIP trunk that support multiple DID. Only the trunk is documented in sip.conf (called DID is taken from the sip-header in real time). I would like to limit the number of simultaneous calls on each DID. Is there a way to achieve this ? My understanding is that the SIP configuration parameter "limitonpeers" will limit at the trunk level, right ? Thanks in advance

For multiline phones how do you set SIP channels to busy. For instance if SIP/101 is on a call then dial would return busy. Right now it just starts ringing on line X, and stacks up from there. What would be really great is if I could control how many calls by the context. So if a call was routed via [overload] Then the ext wouldn't report busy it would just keep ringing available

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

Hello list, I try to setup an asterisk-server with different SIP-Peers to PSTN. The Peer are working and configured in sip.conf: [peer1] type=peer host=10.10.10.1 [peer2] type=peer host=10.10.10.2 Now dialout is no problem. Extensions.conf says: exten => _0Z.,1,Dial(SIP/49${EXTEN:1}@peer1,30) But how can I setup a failure-route if the SIP-Proxy "peer1" ist not

...: Rename option "emailtitle" to "emailsubject" * Voicemail: Add "nextaftercmd" option to voicemail to move to the next message automatically * Voicemail: change to flow of voicemail login when password is bad * pbx: Give Busy() and Congestion() an optional timeout * GroupCount: Allow multiple groups per channel (group categories) CHANNELS * chan_iax2: Add IAX provisioning support to Asterisk * chan_iax2: Change IAX2 channel naming convention * chan_sip: When doing rport, remove the ";rport" before adding ";rport=5060" * chan_sip/rtp: Extend sip.conf...

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales