similar to: Using asterisk as the recording server

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

Trying again... *Summary:* I need to have 2 machines with 4 BRI connections, 2 in NT mode, 2 in TE mode and 1 machines with 2 BRI connections, 1 in NT mode, 1 in TE mode; What card(s) should I put in to these servers? *The long story:* I have 3 locations I want to connect using (*) servers. 1 of those has a single BRI with a Siemens DECT PABX. 1 of those has two BRI's with 2 Siemens DECT

Hi it's possible to use a Digium TE110P Single T1 / E1 PCI Interface for supply a E1 link to a old PABX ? Thanks

Greetings all- I'd like to install Oreka on a Centos 5.x server for monitoring my Asterisk systems(using port mirroring) but I find I'm having problems with the version of libpcap installed. The latest is 0.9.4 but the orkaudio RPM (built for Centos 4.2) requires libpcap 0.8.3. I've tried making symlinks to overcome the issue but with no success. Without having to build Oreka from

Hello folks, I wanted to setup Oreka to monitor calls on a trixbox box I have setup. Oreka doesn't seem to be catching all of the calls though.... I have port mirroring setup on the port that trixbox is connected to, mirrored to the port Oreka is connected to. I have read that Asterisk doesn't work as a SIP Proxy, so I wondered if this meant that some phones, after checking in with

Hi all, My name is Ricardo and unfortunately I'm just crawling in this telecomm/asterisk world. So, after reading all day long i still don't understand a few things. :D I'm trying to "develop" a call recorder for a costumer. He has a small call center ( 10 agents ) and want to record all calls. Since he already has everything (ACD only) working perfectly in the PBX and

Hello; I am looking for calls recording solution to do recording based on the network traffic .. The solution to be competitive and appreciate if it is open source .. Any suggested one? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131021/8ed5196a/attachment.html>

Deal All Asterisk Expert If this possible to Create Voice Recording System Beside Main Asterisk PBX?, so Call be handle by 1 Server and Recording by other server. 1. How to accomplish. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090325/d57465cd/attachment.htm

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call

Thks for your answer, here is an exemple of what i do with the errors in french... > tmp [1] 200 150 245 125 134 345 320 450 678 > beta18 Erreur : Objet "beta18" not found //NORMAL just to show it > eta [1] 500 > func1<-function(beta18) dweibull(tmp[1],beta18,eta) > func1<-func1(beta18) * function(beta18) dweibull(tmp[2],beta18,eta) Erreur dans dweibull(tmp[1],

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

Hi! Anybody familiar with streaming rtp from asterisk. Preferably with the xorcom asterisk patch which streams rtp from asterisk to oreka audio server. Any ideas will do just fine though! Regards / Marcus

Hi All, Can Asterisk dial extension which resides in the PABX? (eg. 2000) Sip Phone <-----> Asterisk <------> ATA (FXS) <------> (CO side) PABX <-----> Extension (eg. 1000) (2100 & 2101) can my sip phone call to pabx extension 1000? What will be my dial plan? I know I can connect to 1000 by

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a

hi all I wish to thanks every body on the R mailing list for answering very fast, directly in my mail box ;). I've finish my work with R and i can say that it is very difficult at the beginning, and when you succeed you are stopped by a stack overflow when you call your nice recursive function (which was working with a tab of 100 element) with a tab of 900 elements, but R just do what you

Hello all! I'm VERY new in using VoIP. I'm looking for any tip or trick to connect a physically PABX behind an Asterisk-System(or similar) via an SIP to Analog- or ISDN-Converter. The point is, I _need_ to deliver calls to extensions in the connected PABX directly (in ISDN-speech "DDI" (DirectDialIn)) without intervention of an operator. Is this technically possible, and if

Hi, We are trying to port over a PABX to our network. Both PRI's seem to be live however, whenever someone dials out from the PABX Asterisk happens to report : -- Extension '' in context 'samsungincoming' from '736327438' does not exist. Rejecting call on channel 0/31, span 2 If crc4 is turned off, it reports a yellow alarm. Any suggestions? Regards, Sahil

Hi I am in the middle of move a small business over from legacy PABX + PSTN lines to VOIP infrastructure. I borrowed a spa9000 to place between the PABX and the PSTN lines. I have had this going for a while (>5 months) and it has been working fine (some issues with echo and other minor things), which is why I am moving to asterisk. I bought a tdm410 with 3 fxo + fxs. The fxs is connected to