similar to: sip show peers returns several notices

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

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

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

The Asterisk Development Team has announced the release of Asterisk 1.6.1.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.12 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix multiple issues with musiconhold, which led to

The Asterisk Development Team has announced the release of Asterisk 1.6.1.12. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.12 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * Fix multiple issues with musiconhold, which led to

Our Samba server running on CentOS 5.1 (upgraded from 5.0) with Samba 3.0.25b-1.el5_1.4 has been in place for about a year and a half. We had no issues till about 2 months ago. This may have coincided with the upgrade of CentOS 5.0 to 5.1 (which would have brought a new Samba version with it). Anyway, the server will be fine and snappy for a week or so, then out of the blue, nobody can

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. 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 tried searching, so if this has already been discussed please point me in the right direction. On separate occasions I've seen cases where Asterisk boxes will be unable to register with each other via SIP or IAX2 when a Firewall is replaced. I'll describe two different cases. In both we have three offices connected via IPsec tunnels. Case 1: High Availability firewall

Please consider the following inputs: PrsnSerialno<-c(735,1147,2019,4131,4131,4217,4629,4822,4822,5979,5979,6128,6128,7004,7004, 7004,7004,7004,7438,7438,9402,9402,9402,10115,10115,11605,12693,12693,12693) PrsnAge<-c(59,48,42,24,24,89,60,43,47,57,56,76,76,66,70,14,7,3,62,62,30,10,7,20,21,50,53,44,29)

With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF} is [un]set in an odd way. for example consider: 999,1,Swift(some long message that you dont want to wait for|5000|5) 999,n,NoOp(DTMF: ${SWIFT_DTMF}) if while I am listening to the playback, i interrupt and dial: - "12345", SWIFT_DTMF is set to

Hello, We're seeing lots of warnings like the following, running Asterisk 1.6.1.12. Does anyone know the cause or cure? One explanation I've come across is that the peer is congested and sending RTCP messages asking us to slow the RTP down. Is there any way we can verify this? [May 17 13:42:45] WARNING[27482] channel.c: Exceptionally long voice queue length queuing to Local/12126412121

Dear R-devel members, I made a small modification in the graphics::barplot function when used with a matrix and beside argument set to false in order to be able to order each bar according to its value (from smaller to bigger or bigger to smaller), while keeping the colors. It may be of general interest (for example to be able to visualize the occurrences of letters from different texts, or the

COnsider the following: Age<-c(48, 57, 56, 76, 76, 66, 70, 14, 7, 3, 62, 62, 30, 10, 7, 53, 44, 29, 46, 47, 15, 13, 84, 77, 26) SerialNo<-c(001147, 005979, 005979, 006128, 006128, 007004, 007004, 007004, 007004, 007004, 007438, 007438,009402,009402, 009402, 012693, 012693, 012693, 014063,014063, 014063, 014063, 014811, 014811,016570) TestSet<-cbind(Age,SerialNo)

I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. On my Aastra 480i phone, I only see the first mailbox listed. I've verified this, by changing mailbox= to reverse the order, and I then see 8150 when I go to Services > Voicemail on the phone. I also only get MWI events for whichever mailbox is listed

I'm playing around with an ALIX 2D2 board (http://www.pcengines.ch/alix2d2.htm). It's a fanless, x86 system using an AMD Geode processor with 256MB of RAM. Also available are two network interfaces, two USB ports and one serial port (no keyboard or VGA). I'm using the Voyage Linux distro, which basically is Debian Lenny optimized for this board. Asterisk 1.6.1.12 runs fine on the