similar to: Dialplan Gurus? Can Asterisk 1.4x CHANNEL function be used to retrieve info about OTHER channels?

Hi, I'm trying to access audionativeformat / other codec variables in the hangup handler of a call (with ${CHANNEL(audioreadformat)}), but I get no response. Also 'core show channel ...' doesn't list those variables. Are they always set by asterisk, or only in some scenarios? It's a simple SIP-SIP call with audio passing through asterisk, same codecs on both sides. I see that

Hello folks, i installed asterisk 1.8 from repo: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages And Looked at this article about CDR in mysl. http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql I installed asterisk-mysql pacakge from debian repo. The cdr in mysql is working, but i can not get cdr adaptive colums are not, i use this in my extension.conf exten =>

Hi all, forgive the question but is it possible to use PCI modem cards (aka winmodem's) as FXO/FXS ports in * ? what about external modems like the USR Sportsters? Thanks in advance, Bryan. Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au

Hi Guys, I seem to not be able to find any good open source Asterisk Queue Analyzer and Asterisk Log Analyzer on the web. The Asterisk Queue Analyzer is to serve as the graphic tool for call center or pbx admins. It will pull the info in queue.log and in MySQL asterisk CDR to create a graphic bar or to report on each extension that received the queue calls, etc... The Asterisk Log Analyzer is

Hi, I am trying to find a way to add the following info in CDRs (with asterisk 1.4.23.1): 1. Codec used 2. RTP QoS statistics 3. RTP IP of remote host 4. For answered calls, the peer that requested to end the conversation I have managed to get 1 and 2 for the caller, like that: exten => h,1,Set(CDR(userfield)=RIP=${SIPCHANINFO(recvip)}

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

I have a high volume asterisk 1.40 installation and I ran a SIP SHOW CHANNELS. (see partial output below). My questions are: 1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls? (2 channels per call) 2. The latter part of the output shows "unkn" for Form column. Why does it not know the codec? Could it be UDPTL? Or are these calls messed up? 3.

Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16.

Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : "sip show channels" [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels" Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0

Hi, We're holding ourselves back from moving to PJSIP as we don't appear to have figured out how to force codec preference in a dial plan. The 'PJSIP Advanced Codec Negotiation' document (https://wiki.asterisk.org/wiki/display/AST/PJSIP+Advanced+Codec+Negotiation) appears to ultimately be what we're after, but we're not comfortable running Asterisk 18 in production just

Hi Guys, I have a channel that is dialed with *Timeout* option. So, there is definite time to it. Only thing is that I don't have control of that channel. I only know that it's using g729 codec and that there is only one channel that is using g729 at any given time. So, my question is:

Hi Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can

Hi, there are a few questions posted on the new Samba forum on Tek-Tips web site http://www.tek-tips.com/threadminder.cfm?SPID=865&CFID=34009252&CFTOKEN=12959894 which I am sure would be easily answered by some of the gurus on this list. You may be asking - "Why don't they just post the questions on this list?" Certainly, they could, but having used the Tek-Tips forum for

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Hi, Is there a variable that contains the Sip Peer name? I was using ${CALLERID(num)} for outgoing calls, but when a call is being transferred, that variable contains something else. I need a variable that is always set to the SIP Peer's name. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Did you check out that iPhone SDK keynote? It makes me want to sell software. Do any of you GNU LGPL gurus understand the license http://www.winehq.org/site/license to know if we can distribute Wine to run our MS Windows executable? I'm sure it's something basic (for a license groker). I'm not linking with Wine, I just want to install it on the iPhone (if Apple allows it through

I''ve been stuck on this particular problem for quite some time now. If I don''t explain it well enough I''d be happy to explain more. I have a group of draggables and a droppable object. I''ve created a function for the Droppable.onDrop event so when I drop one of these items I check (via an array) to see if it has already been added to the droppable. If it

> Looks like this is an AMD/ATI driver issue. Not really. I think you have missed the previous posts. It's the exact same problem with NVIDIA. And VirtualBox too. It seems Macs with NVIDIA can go a little bit more far, but they crash too. It's surely a wined3d issue.

SolisX wrote: > Hello JHaleIT,m > > Just tried it with wine 1.3.37 and it crashed at just before loading the ship lobby screen. > So it must be something wrong with your installation of wine or maybe your distro. I'm using wineskin. I tried creating a new wrapper as well with no luck. It just freezes and shows a black box on the left corner of my screen. I have to force quit it.