similar to: Get CONNECTEDLINE info from other Asterisk system via IAX2

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 all connected line info. Looking at a pcap trace of the D-channel data, I

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:

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 ----------------> Asterisk ------------------->Bob ------- INVITE

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 Line 955" when my call is answered, shouldn't the

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!

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})

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

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

>>> 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

This works, where zz is a dataframe: for(i in 1:nrow(zz)) { zzz[i,1]<-paste(zz[i,1],zz[i,2],sep="_") } I would like to use "apply" to concatentate two columns of text along with a separator. How? Chet [[alternative HTML version deleted]]

Hello everybody, from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess. Apart from logging all traffic 24/7 via tcpdump (not really

This doesn't work, what should I do? sudo apt-get install r-base [sudo] password for cseligman: Reading package lists... Done Building dependency tree Reading state information... Done r-base is already the newest version. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. Also: > .libPaths() [1] "/export/home/cseligman/local/library/R" [2]

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

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 1.6.1.20. What do i

Hi Peter, Martin, and others, Thanks for your replies. - The bugs apply to all systems that use GNU Readline, not just Linux or Arch Linux. - Readline version 6.3 changed the signal handling so that SIGWINCH is no longer handled automatically by the library. This means it's not currently possible for people using R on e.g. Linux to resize the terminal, or at least when they do so

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

Hello: Ed Zaleski ran some remarkable and very impressive load tests on icecast reported on www.icecast.org (Home page). What wasn't mentioned in the test specifications was the type of internet line that was used which totally influences bandwidth and subesequent overall load characteristics. Questions: 1. Was it a T1 line, Verizon FIOS, or cable internet, or what? 2. What were the

Hello all, I need to configure a new provider to complete calls to us, the provider gave to me 2 different ip address, one is the default host and another to RTP server, so far as i knew the rtp server should be the same address but different ports, anyway i think i?m completelly wrong about it.. someone could tell me how can i configure in asterisk this connection in sip.conf? Thanks, Chet

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

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP