similar to: How to tell if asterisk is handling media or not?

Hi all, This issue is giving me a lot of grief with my customers. I have 5 asterisk servers running in production, each one with almost 70 simultaneous calls at peak hour. Most of my customers complain that their calls drop after 20 seconds or so. After running through my cdr's, I see that the number of 20 second calls is MUCH larger than any other number. (see below) billsec count(*) 1 924

Much to my surprise I tried to debug an AGI script today with "agi debug" on the Asterisk CLI and it did not work. Plus, I could find no reference on lie of it being removed. Is there another name for that command? I scanned the CLI help but found nothing similar. Both my 1.6 boxes do not have the command but my 1.4 box does. Thanks! Alex

Hi all, I am having a curious problem. I use two cdr backends, csv and MySQL. These are my settings: Call Detail Record (CDR) settings ---------------------------------- Logging: Enabled Mode: Batch Log unanswered calls: Yes * Batch Mode Settings ------------------- Safe shutdown: Enabled Threading model:

Hi all, I've been trying to add a custom mysql field to my CDR's, but I must be doing something wrong. I am using asterisk 1.4 and asterisk 1.6, in extensions.conf I add: exten => h,1,Set(CDR(q931)=${HANGUPCAUSE}) This extension is executed, I can see it in the asterisk console. I have added a new column in my MySQL database called q931. However, the new field does not show up in

Hello, I'm wondering if anyone else has been observing problems lately with 1.4.18 and higher releases of asterisk not properly recording calls. When using MixMonitor, the resulting file is only a few bytes long. I think this is because asterisk is doing Native bridging even though MixMonitor should block that. Did something change around the release of 1.4.18 that would have changed

Hello everybody. I have a provider that has 3 asterisk boxes which I must balance my calls against. At the moment, I route different destinations to different boxes but this causes lots of problems. Without resorting to OpenSER or other proxies (as my provider also uses IAX), is there a way I can load balance outgoing channels in Asterisk? For example an IAX peer like: [iax_provider] type=peer

Hello all, I have a pretty much standard installation of an Asterisk 1.6.1.6 with no PRI cards of any type (full VoIP). Occasionally (it happens every 2 weeks or so), it just stops running. I was using safe_asterisk but it seems that safe_asterisk did not restart it. I do have the core dump file at /tmp/core.myservername-2009-10-20T18:36:20+0200 but it seems it's from an earlier crash. When

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

Dear All Can you please do me favor and let me know what is the equivalent of "iftop" command on CentOS 5.2 ? Please be informed that I couldn't find an rpm package for "iftop" on the www.pbone.net . Let me thank you in advance -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to get a package (iftop) installed on a test node using puppet but I can''t get it to install. This seems like it should be simple, but I can''t figure out what I am missing. Have done a lot of searching for similar problems and the answers I found seem to be vague in the details and explanation. I have the package added to an internal yum repo. OS is Centos 5.4.

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

Hi, I put reinvite=yes in my sip.conf. For testing, I restricted the codecs to alaw. I have no modifiers in my dial command. Thus, there should be no reason not to reinvite. Call (sip, authenticated) comes in and is forward via SIP (not authenticated) to another asterisk box. Unfortunately, media path still passes through the asterisk box in the middle. Using sip debug I even can't find

Hi all, I am having trouble getting cdr_adaptive_odbc to work. I have correctly configured the odbc drivers and dsn (I have tested this by connecting directly using isql). I have also configured /etc/asterisk/cdr_adaptive_odbc.conf like so: [test-asterisk] connection=test-asterisk-odbc table=cdr I have tested the ODBC connection test-asterisk-odbc and it works correctly However when I try to

I am currently suffering various SIP attacks. I am using the following extension to record the caller's IP address: exten => h,n,set(CDR(srcip)=${CHANNEL(recvip)}) However, in recent attacks, this IP address is not correct, and I believe that they are spoofing it. I am using asterisk 1.6.2.15. Does the CHANNEL(recvip) variable record IP show in the SIP header instead of the real, UDP

Hello everybody, When I execute the "sip show peers" command in the asterisk console I always get the following notice, repeated 15 times after the sip show peers output. [Dec 21 03:38:31] NOTICE[12693]: utils.c:1074 ast_wait_for_output: Timed out trying to write This happens on a freshly installed 1.6.1.12 and a 1.6.1.6 box that I am running. Both of them use Debian Linux (lenny) on

Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood