search for: outbound_group

...macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource. snom300-for-vasya ; ${ARG3} - dial timeout ; ${ARG4} - dial options ; ${ARG5} - dial url exten => s,1,Goto(s-set-variables,1) exten => s,n(set_var_ret),Set(GROUP(${LOCAL_PARTY})=OUTBOUND_GROUP) exten => s,n,GotoIf($[${GROUP_COUNT(OUTBOUND_GROUP@${LOCAL_PARTY})} > 1]?play_back_busy) exten => s,n,GotoIf($[${LEN(${CALLERIDNUM})} != 4]?skip_down) exten => s,n,Set(GROUP(${CALLERIDNUM})=OUTBOUND_GROUP) exten => s,n(skip_down),Noop("Test") exten => s,n,NoOp(${OUTBOU...

Dear all, Anyone has experience using group and group_count to limit outgoing calls in AGI/CAGI? SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP EXEC Gotoif $[${GROUP_COUNT(OUTBOUND_GROUP@${CALLERIDNUM})} > 1]?BLOCK SET VARIABLE GROUP(${CALLERIDNUM}) OUTBOUND_GROUP But it doesn't work as it should. Tried in extensions.conf and it works. Any idea. Thanks Ray -------------- next part -------------- An HTM...

...n 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 "OUTBOUND_GROUP") at pbx.c:5904 #2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0, peerflags=0xf469fee8) at app_dial.c:964 #3 0xf5bc23ed in dial_exec (chan=0x0, data=0x1) at app_dial.c:1601 #4 0x08090aed in pbx_extension_helper (c=0xa281820, con=0x0, context=0xa281970 "default", exten=0xa...

...ehaves this way even if the users' voicemail context is something else, so I am guessing that is hard-coded in FOP somewhere.) If I log them in from Local/${AGENT_EXTEN}@default, FOP works and the agents get the calls, but then it's just dialing them directly - there is no way to increment OUTBOUND_GROUP or check the value of GROUP_COUNT. As a result, calls are routinely sent to agents who are already on the phone, which I don't want. Obviously, the next reasonable solution would be to use some other context for the default context, and use [default] instead of [agents] for incrementing OUTBO...

Hi, i have diffirent provider example(3 single account in deltathree, 4 account in packet8 and so on) . How this possible to make the three individual sip account in deltathree act as trunk so that i cannot get a busy call. If line one fail goto line 2 then line 3 or another trunk line 1 then line 2 then line3....I read it in asterisk at home but the script i am copying is not working .

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate

...is processed when a queue dials 611 looks like: _XXX => { // internal dial to extensions from queue. Set(GROUP()=${EXTEN}); // increment group count Set(CDR(accountcode)=ext${ext}); // for Phoenix Set(OUTBOUND_GROUP=${EXTEN}); // same for channel that will be created by Dial() NoOp(GROUP_COUNT of ${EXTEN}: ${GROUP_COUNT(${EXTEN})}); if ("${GROUP_COUNT(${EXTEN})}" = "1") // if not already in call { Set(DIALST...

Hi All, I use the SetGroup command to identify if a specific extension is in use. I create a group for each extension and check against that group name when putting through any further calls. A problem I am finding is that with internal calls I want to increment both the called and calling extension and SetGroup only appears to allow a call to be in a single group. Ideally I would like to

Hi, i'm new in asterisk programming. Maybe my question was posted thousand times but i found nothing using google. I'm looking for a method to limit the total simultaneous calls (inbound and outbound) that pass from internal phones to 2 SIP providers. I found the calllimit option but it works only on a per-channel basis. Instead i want limit the total amount of calls, abstracting from

...example SIP/102) callagent is macro which dials phone where agent is. The shortened version of this macro is: [macro-callagent] ;ARG1 - What device to dial (device) ;MACRO_EXTEN is dialed agent exten => s,1,Gotoif($[${GROUP_COUNT(${MACRO_EXTEN}@agents)} > 0]?busyhere,1) exten => s,n,Set(OUTBOUND_GROUP=${MACRO_EXTEN}@agents) exten => s,n,Monitor(wav|/var/calls/${UNIQUEID}) exten => s,n,Dial(${ARG1},${AGENT_TIMEOUT},${DIAL_OPTIONS}) exten => s,n,Busy As you can see I would like to monitor all calls to the agents. That's where the problem is. If we use Local channel without /n option...

I have a rather long dial plan, but it includes support for call waiting. However, the setgroup checkgroup commands don't seem to be working. Can anyone help on this one? Excerpts are below. First exten-vm is dialed and then dial-new. As I understand, priority 1 increments the active channels for the caller and then in "dial-new" priority 8 increments for Arg3, or the Callee

I want to limit the number of simultaneous incoming calls that my IAX DID can accept to, say, 2. The IAX DID provider sets no limit. The code below does work, but when the limit is in effect, new callers hear a "call cannot be completed as dialed.." message instead of a busy signal. Maybe this is an issue with the provider, but I do not like this and want callers to hear a busy signal.

Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian

...Set("SIP/100-b609f9c0", "DIAL_NUMBER=505103150") in new stack -- Executing [s at macro-dialout-trunk:5] Set("SIP/100-b609f9c0", "DIAL_TRUNK_OPTIONS=trf") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/100-b609f9c0", "OUTBOUND_GROUP=OUT_12") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/100-b609f9c0", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/100-b609f9c0", "0?skipoutcid") in ne...

...ot;, > "DIAL_NUMBER=015921256331") in new stack > -- Executing [s at macro-dialout-trunk:5] Set("SIP/75002-b7705298", > "DIAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [s at macro-dialout-trunk:6] Set("SIP/75002-b7705298", > "OUTBOUND_GROUP=OUT_7") in new stack > -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/75002-b7705298", > "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [s at macro-dialout-trunk:9] GotoIf("SIP/75002-b7705298", > "...

...;DIAL_NUMBER=15921256331") in new stack > -- Executing [s at macro-dialout-trunk:5] > Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Ttr") in new stack > -- Executing [s at macro-dialout-trunk:6] > Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack > -- Executing [s at macro-dialout-trunk:7] > GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack > -- Goto (macro-dialout-trunk,s,9) > -- Executing [s at macro-dialout-trunk:9] > GotoIf("SIP/192.168.0.151-088e79...

Hello everyone! I've had this problem for a while and cant figure it out. When an outside caller calls an extension on my asterisk system, they do not hear any sort of ringing. Inside extensions calling other extensions do hear ringing. We have 3 other asterisk systems that are configured the same way, but do not have this problem. We think it has something to do with asterisk 1.6. The other

Hi this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

...w stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:5] Set("SIP/21-00000058", "DIAL_TRUNK_OPTIONS=tr") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:6] Set("SIP/21-00000058", "OUTBOUND_GROUP=OUT_2") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/21-00000058", "1?nomax") in new stack [Apr 30 09:34:31] VERBOSE[12649] pbx.c: -- Goto (macro-dialout-trunk,s,9) [Apr 30 09:34:31] VERBOSE[12649] pbx.c:...

...Set("SIP/101-00000103", "DIAL_NUMBER=0033149xxxxxx") in new stack -- Executing [s at macro-dialout-trunk:5] Set("SIP/101-00000103", "DIAL_TRUNK_OPTIONS=tr") in new stack -- Executing [s at macro-dialout-trunk:6] Set("SIP/101-00000103", "OUTBOUND_GROUP=OUT_5") in new stack -- Executing [s at macro-dialout-trunk:7] GotoIf("SIP/101-00000103", "0?nomax") in new stack -- Executing [s at macro-dialout-trunk:8] GotoIf("SIP/101-00000103", "0?chanfull") in new stack -- Executing [s at macro-dialout...