search for: dialgroup

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

...}_in,mb) exten => 425,n,Dial(${PHONE1},10) Now, I want to create a call group: I mean, I want a number (eg 800) that makes 3,4...10 phones ringing together. I found 2 modes to do this: 1) exten => 800,n,Dial(${PHONE1}&${PHONE2}&${...},15) 2) with Asterisk 1.6: exten => 800,n,Set(DIALGROUP(test,add)=${PHONE1}) exten => 800,n,Set(DIALGROUP(test,add)=${PHONE2}) .... exten => 800,n,Dial(${DIALGROUP(test)}) How can I record a call made to the number 800 but that will be stored on the directory of the phone (eg ${PHONE2}) that really picks up the call? Thanks a lot in advance,...

...r(Alert-Info: <http://127.0.0.1>\;info=alert-internal) .. Dail(SIP/2010) But still have the phones all ring in parallel. Any and all suggestions welcome. With kind regards, Dw. [sub-callout] exten => s, 1,Log(NOTICE, Ringing ${ARG2}) same => n,Set(DIALGROUP(allgroup)=) same => n,While($[${LEN(${ARG1})} != 0]) same => n,Set(PEERNAME=${SHIFT(ARG1,&)}) ; Skip any peer that is currently not connected same => n,ExecIf($[0${SIPPEER(${PEERNAME},port)}...

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

...defined: [options] [...] defaultlanguage= de_DE In extensions.conf, I use: [general] format= gsm [...] and for the inbound endpoint: [inbound] exten => THENUMBER,1,SetGlobalVar(CHANNEL(tonezone)=de) same => n,SetGlobalVar(CHANNEL(language)=de_DE) same => n,Set(DIALGROUP(group_isp2,add) = PJSIP/502) same => n,Set(DIALGROUP(group_isp2,add) = PJSIP/512) same => n,Dial(${DIALGROUP(group_isp2)},30) same => n,VoiceMail(502 at my-vmmbox) same => n,Hangup() and in voicemail.conf I have [general] [...] tz= german...

...ly use a P2P trunk, but if there are no free channels the system tries a separate P2MP trunk. In case the congestion is caused by the called party, switching to another trunks does not make any sense, so I need to find out whether my side is causing the CONGESTION. Has somebody tried to setup a dialgroup where P2P, P2MP, and POTS devices are all part of the same group? This would also solve my problem. In chan_dahdi.conf there would be something like context=from-pstn-p2p group=2 signalling=bri_cpe channel =>1-2 context=from-pstn-p2p group=2 signalling=bri_cpe channel =>4-5 context=from-...

Hi List, Merlin Magix hardware v02 I'm trying to get asterisk to act as a voicemail server for a lucent merlin magix PBX that we purchased used. We have 4 FXO channels between the two PBXs on a Sangoma A200 card. The 770 dialgroup is working properly, in that calls to 770 are answered by Asterisk. The magix is sending mode codes in the format #XX#XXX#, where the 2nd block of digits is the calling extension. I'm stripping off the unneeded pound signs and digits, and calling voicemailmain. The problem I'm having...

Greetings. We are running * on RH9 using a Digium TDM400P four-port FXO card. We use only two ports on the card (ports 0 and 2 in this case). A consultant set this up for us, and it mostly works OK. However, outbound calls use our secondary number rather than our primary number first. This undesirable because of Caller ID; we'd like the primary number to appear instead. Yes, I can

...g/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve duplicated data in the AstDB when using DIALGROUP() (Closes issue #18091. Reported by bunny. Patched by tilghman) * Correct issue where res_config_odbc could populate fields with invalid data. (Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman) * When using cdr_pgs...

....org/pub/telephony/asterisk/ The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve duplicated data in the AstDB when using DIALGROUP() (Closes issue #18091. Reported by bunny. Patched by tilghman) * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (Closes issue #18464. Reported, patched by IgorG) * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for...

...g/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve duplicated data in the AstDB when using DIALGROUP() (Closes issue #18091. Reported by bunny. Patched by tilghman) * Correct issue where res_config_odbc could populate fields with invalid data. (Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev, jthurman, elguero, zerohalo. Patched by tilghman) * When using cdr_pgs...

....org/pub/telephony/asterisk/ The release of Asterisk 1.8.3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * Resolve duplicated data in the AstDB when using DIALGROUP() (Closes issue #18091. Reported by bunny. Patched by tilghman) * Ensure the ipaddr field in realtime is large enough to handle IPv6 addresses. (Closes issue #18464. Reported, patched by IgorG) * Reworking parsing of mwi => lines to resolve a segfault. Also add a set of unit tests for...

In my * box I've configured two queues and incoming number and whenever any one calls those number call comes to my *box and it sends call to my agents in queue. but if no agent is available it still answer the call. Is there any why when my agents are not available I don't want call to get answered. Here is my dialplan: exten => xxxx,1,GotoIfTime(*|*|20|dec?ccagents,xxxx,6) exten

Hi List, I have faced a problem in asterisk queue implementation. I configured a queue with 'ringall' strategy and 'ringinuse=yes' in queues.conf. If three calls come to this queue in parallel, the logged in queue agent used to get only one call (may be the first one), not all the calls waiting in the queue at a time. Once the agent answers the call the next call is displayed. I

Hello I have the following scenario: [mynicecontext] exten => 2000,1,Dial(SIP/deviceA&SIP/deviceB&SIP/deviceC) As expected, by dialing 2000, all three devices will ring. And that's fine. However, there are situations where I only want "deviceA" and "deviceB" to ring. I would like to have an extension to dial in order to modify the dialplan. Here is what I