search for: callername

.... _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on Polycom 500 shows CALLERNAME only Does anyone know how to change the display format of caller id on the screen of a polycom 300/500/600? When I call FROM my 'shop phone 203' TO my 'office phone 201', a Polycom 500, it only says 'Shop' as the calling party. More specifically, the two lines look li...

...al Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Eric Wieling Sent: Tuesday, August 31, 2004 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP 300 - Displaying Only CallerNAME... What about NUMBER? Put a NoOp(Caller*ID is ${CALLERID}) in your dialplan JUST before the Dial to the Polycom. See if the correct name and number shows up on the console when the NoOp runs. If it does, there's a problem in the Polycom, if there is no NAME then you have a problem with your...

...pPolicy>3</messageWaitingLampPolicy> <messagesNumber>*97</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>7b452e87-4496-4762-e11f-b26751a1884b</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> <line button="2"> <featureID>9</featureID> <fe...

...<messagesNumber>400</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <contact>study_line1</contact> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>true</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> <...

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi there, I want to add 4000 Callerids and Callernames to my asterisk-db. (/var/lib/asterisk/astdb) I do not want an external database or an sql-database because I do not want asterisk to depend on external processes. However, when I do 4000 "database put number name" via a shellscript and "asterisk -rx" I only have 600 entries...

...nk on their response. And since this isn't a vmail account and trying to avoid an AGI script if possible I'm not sure how to email the recording(s). I also want to be able to structure the body of the email so that it reads something like You have a new call from $CallerID - "$CallerName" on 'DateTime' ... ect, ect. James Shigley Monroe Telephone Answering Service 409-981-9213 Infinity 5.5,UC 4.02.3803, Blink 3.0.104 Ecreator:2.21, eResponse 1.1.7 Webportal,WebApps, CONFIDENTIALITY NOTICE: This email, including any attachments, contains information which m...

...at the SPA is registered as an extension on my system, and incoming calls are routed into the system VIA that extension. The dialplan that the SPA connects to is: [gw8028] exten => 8028,1,Answer exten => 8028,n,Set(CallerNum=${CALLERID(num)}) exten => 8028,n,Set(CallerName=${CALLERID(name)}) exten => 8028,n,Set(CDR(accountcode)="8203") exten => 8028,n,Set(CDR(UserField)="POTS") exten => 8028,n,Goto(from-internal,111,1) exten => 8028,n,Hangup the 'from-internal' is my current call filtering/proc...

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Hello all, I'm new to asterisk. Our company wants to setup an asterisk server and will eventually move to IP centric phones, but they don't want to just throw away the old Siemens PBX, so during the process we want to integrate it with asterisk. Is it possible? and how? thanks. Lito -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have the 7970 xml config for sip yet? Aaron

...} $agivar = explode(':', $agivar); $agivars[$agivar[0]] = trim($agivar[1]); } extract($agivars); // Variable Declarations $agi_uniqueid; $agi_callerid; $agi_calleridname; $agi_extension; $agi_uniqueid; $UNIQUEID = $agi_uniqueid; $CALLERID = $agi_callerid; $CallerName = $agi_calleridname; $EXTEN = $agi_extension; $Q1 = "Did Not Answer"; $Q2 = "Did Not Answer"; $Q3 = "Did Not Answer"; $Q4 = "Did Not Answer"; $Q5 = "Did Not Answer"; // Q1 fwrite(STDOUT, "STREAM FILE /var/lib/asterisk/sounds/1 ###\n&...

...yone know an interesting website where I can find infos about UUI in >> ISDN (with CAPI maybe?) ? > >I guess it's somewhere in ITU Q.931, but i dont have this document ;-( > >I also think this would be a very cool feature (i.e. there's a Simemens PBX >that sends out the callername with UUS1), if i can do something else to help, >please tell me. > > >Regards > >Christoph

Hello All, I was wondering. What are the primary advantages to using a PRI over a T1? As I understand it, the PRI terminates very fast, meaning you can do immediate answer and dial... This is very handy on the BRI line I have on the asterisk. Can T1 signalling also do immediate answer, or does it just behave like a channelized pots line and ring as usual? I am trying to determine if I should

...ERID(all)=<450-655-****>) exten => _9XXXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) And here's a Macro that I use for incoming call for VoiceGenie: [macro-voicegenie] exten => s,1,Answer exten => s,2,SIPAddHeader(X-Asterisk-DID: ${ARG1}) exten => s,3,SIPAddHeader(X-Asterisk-CallerName: ${ARG2}) exten => s,4,Dial(SIP/108) exten => 514380****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)}) exten => 514380****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)}) exten => 514373****,1,Macro(voicegenie,${EXTEN},${CALLERID(name)}) exten => 514373****,1,Macro(voicegenie,${EXTE...

...2-1", "dtfb|GET|CallerLevel|"SELECT caller_level FROM callers WHERE caller_phone = ''"") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dtfb -- AGI Script dtfb completed, returning 0 -- Executing AGI("Zap/2-1", "dtfb|GET|CallerName|"SELECT caller_name FROM callers WHERE caller_phone = ''"") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dtfb -- AGI Script dtfb completed, returning 0 -- Executing NoOp("Zap/2-1", "Caller Level is ") in new stack -- E...

[Reposting, as was bounced for non-member, sorry if this is a dupe] Arrangement: { PSTN }--E1--[PBX]--E1--[*]--LAN--[SIP phones] \__[PBX system phones] Normal calls between PBX system phones and SIP phones work, in both directions. The call logs look like (ignore the no answer, it did ring):

Hello guys, We're currently working on asterisk trying to create our own SIP phone, because we need special features. But dunno maybe there's other people who already done that before. Basically, we are a inbound call center. We have serveral customers with different phone numbers, which are redirected to us. When we receive a call coming on a specific phone number, the call gets

...; exten => s,1,Answer() exten => s,n,MYSQL(Connect connid localhost asterisk xxxxxx yyyyy) exten => s,n,MYSQL(Query resultid ${connid} SELECT channel, extension, name FROM pbx WHERE cid_num='${MACRO_EXTEN:4}') exten => s,n,MYSQL(Fetch fetchid ${resultid} channelpath CALLBACKNUM callername) exten => s,n,MYSQL(Clear ${resultid}) exten => s,n,MYSQL(Disconnect ${connid}) exten => s,n,AddQueueMember(${ARG1},SIP/${MACRO_EXTEN:4}) ;If they're already logged in, log off exten => s,n,GotoIf($["${AQMSTATUS}" = "MEMBERALREADY"]?out) exten => s,n,Set(DEVI...