search for: connectedline

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle a...

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.c...

Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice -------------...

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Li...

2015-04-30 17:45 GMT+02:00 Richard Mudgett <rmudgett at digium.com>: > > > On Thu, Apr 30, 2015 at 4:50 AM, Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with >> a couple of SIP phones. >> >> When a SIP phone dials an other one, with a CONNECTEDLINE statement in >> its dialplan, I noticed that Asterisk update caller's information using a >> Remote-Party-ID header in 180 Ringing...

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing an Answer...

Hello! I have following connections over isdn pri: avaya definity---pri--asterisk--pri-panasonic 500 Just because panasonic 500 can't send user's names. I also want to have reverse callerid for avaya users. But if there is no answer in dial plan: exten => _XXXX,1,Set(CONNECTEDLINE(name)=${DB(names/${EXTEN})}) ;exten => _XXXX,n,Answer exten => _XXXX,n,Dial(DAHDI/g4/${EXTEN}) exten => _XXXX,n,Hangup there is no name on avaya display after panasonic user pick up... With answer name appears immediately but this is not what we want :-( Is it possible to do something?...

I have been racking my brain attempting to get the remote callerid information for calls made to extensions on another Asterisk system connected via IAX2 but nothing has worked. To clarify, I would like to display the number AND name on the calling phone when calling extensions on another Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but

I don't expect my SIP provider to provide useful "Remote-Party-ID" information. Therefore, I am using "CONNECTEDLINE(num)=xxx" AND "CONNECTEDLINE(name)=yyy" to populate remote party information from a local database. I am also using the "I" (upper case "i") option for Dial. Generally I like to see to see the remote party name appear on the phone's display as soon as a call i...

Hi, i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)}) Set(CONNECTEDLINE(num)=${EXTEN}) ends with [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6....

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying

On 2020-05-28 11:10, Doug Lytle wrote: >>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... > > And that we don't. > > It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE wh...

Hello Le 28/05/2020 à 21:04, C.Maj a écrit : > On 2020-05-28 11:10, Doug Lytle wrote: >>>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... >> And that we don't. >> >> It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLI...

...[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk:18] GotoIf("SIP/222-00000004", "0?bypass,1") in new stack [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk:19] ExecIf("SIP/222-00000004", "1?Set(CONNECTEDLINE(num,i)=0912345678)") in new stack [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c: -- Executing [s at macro-dialout-trunk:20] ExecIf("SIP/222-00000004", "1?Set(CONNECTEDLINE(name,i)=CID:222)") in new stack [2014-09-23 21:18:46] VERBOSE[11634][C-0000000d] pbx.c:...

...y are doing something similar and maybe I can avoid re-inventing the wheel. > > All comments are welcome! > > Doug > Push messaging with curl is nice for broadcasting text messages to idle phones. But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... same => n,Set(CONNECTEDLINE(name)=User is Busy) ...sort of 'Reverse Caller ID Name' to immediately change what the caller sees on their phone display. More involved is the CALLCOMPLETION function eg. for automating redials to the busy user when they hang up th...

...ion, the callerid of the dialed extension is returned back on the display of the calling phone. So if you call extension 100, you can see you are calling Ann (for example). I want to selectively disable the transmission of this information back to the caller. How can I do it? I tried setting Set(CONNECTEDLINE(num-pres)=prohib); but it doesn't seem to sort any effect. Where am I wrong? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131004/b547c568/attachment.html>

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

...clr:2] Set("SIP/1010-00000161", "GOSUB_RETVAL=") in new stack -- Executing [s at macro-blkvm-clr:3] MacroExit("SIP/1010-00000161", "") in new stack -- Executing [s at macro-auto-blkvm:3] ExecIf("SIP/1010-00000161", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=1010)") in new stack -- Executing [s at macro-auto-blkvm:4] ExecIf("SIP/1010-00000161", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Josh Hopkins)") in new stack -- Feature Found: apprecord exten: apprecord -- Executing [s at macro-one-touch-record:1] ExecI...

...{l_DialTo}@proxy_${l_Proxy},30,b(dialer-header^s^1)G(dialer- playmsg^s^1)) [dialer-header] exten => s,1,Set(PJSIP_HEADER(add,X-Carrier)=${l_Carrier}) same => n,Set(PJSIP_HEADER(add,X-CallerID)=${l_CallerIDnum}) same => n,NoOp(X-Carrier = ${PJSIP_HEADER(read,X-Carrier)}) same => n,Set(CONNECTEDLINE(number,i)=vap_002) same => n,DumpChan(1) same => n,Return() [dialer-playmsg] exten => s,1,Goto(hold,1) same => n,NoOp(Enter Play Message) same => n,NoOp(Path = ${g_pmtPath}) same => n,SayAlpha(${g_SIPUser}) same => n,BackGround(${g_pmtPath}Intro) same => n,WaitExten(60)...