search for: recarey

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

I'm trying to get my asterisk server to reinvite. I have two asterisk servers with public IP's. My users (behind NAT) register on one server (I'll call it server 1), and for some calls they are transfered over to the other server (server 2), because that server has the E1's. I want server 1 to be in the signaling path for billing reasons, but handling the media stream is killing

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

...al Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Doug Sent: Monday, April 19, 2010 9:12 PM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Calls drop after 20 seconds At 20:07 4/19/2010, Alejandro Recarey wrote: >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. Many people...

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

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

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

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

Hi all. I am having lots of trouble with random calls dropping after 20 seconds, and I finally managed to capture a full sip trace. I'll paste it in full below, but I'll give a summary first. It seems that Asterisk is not recognizing the ACK messages that it receives from the Grandstream ATA. This happens only on the ACK that follows the OK that marks a call as established. This makes

Hi all, I am worried because on my production asterisk servers, I am receiving these errors every 2-3 minutes. my log files are full of them: WARNING[xxx] app_dial.c: Unable to forward voice or dtmf and also, less frequent: WARNING[xxx] app_dial.c: Unable to write frame How can I find out what is causing this problem? If anybody can point me in the right direction I would be very grateful.

Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some calls the volume suddenly drops to inaudible o completely fades away for a time. If you hold on long enough (20 to 30 seconds) the sound will come back. My asterisk server is on a public IP, and basically acts as a VoIP bridge receiving calls from my customers (all of whom use Grandstream GXW400X