similar to: peer IP address in CDR

Hi, I have trouble re-registering a G729 license for Asterisk (bought 6 years ago) My license looks like: 10D2XXXXX-XXXX-XXXX-XXXX-XXXXXXXXXXXXX Tried to re-register the codec according to the http://downloads.digium.com/pub/telephony/codec_g729/README document, but the register failed with this error message: You selected 5, G.729 Codec Please enter your Key-ID:

Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-----[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from

Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from '"xxxx" <sip:xxx at xxx.xxx.xxx.xxx>' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message

Hi all, We encountered a problem by chance today, a kernel bug which causes the following error: Assertion failure in journal_fl sh() at journal.c:1310: "!journal->j_running_transaction" kernel BUG at journal.c:1310! This happened while remounting an ext3 filesystem read-only. We often do this and haven't seen this bug before, so I suspect some kind of race. It

If you are running ntop on 5.4, what compile options? Use ports version? Or surgefile tarball? It makes a great security forensics tools, but I can't get it to stop segfaulting.Was wondering if anyone found a fix for it. -- Michael Scheidell, CTO 561-999-5000, ext 1131 SECNAP Network Security Corporation Take a vacation from spam: http://www.spammertrap.com

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi all, I apologise in advance if this is off-topic or not permitted on this mailing list. We are looking to hire a skilled developer as soon as possible. We are looking for someone with the following skills: - Linux kernel development, especially Netfilter and ebtables modules - Perl software development, including DBI - SQL database programming, especially MySQL Although we would prefer to

Hi, I'm evaluating what could keep me from upgrading production systems to 1.6.2. As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I met an issue with BLF-pickup which kept me from going further. Have you met other issues I should include include in my checklist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 ========================================================== == == Subject: Stack buffer overflow in nmbd's logon == request processing. == == CVE ID#: CVE-2007-4572 == == Versions: Samba 3.0.0 - 3.0.26a (inclusive) == == Summary: Processing of specially crafted GETDC == mailslot requests can result in a

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 ========================================================== == == Subject: Stack buffer overflow in nmbd's logon == request processing. == == CVE ID#: CVE-2007-4572 == == Versions: Samba 3.0.0 - 3.0.26a (inclusive) == == Summary: Processing of specially crafted GETDC == mailslot requests can result in a

Hi all, We would like to increase the size (sample length) of RTP packets sent by Asterisk to SIP phones. I gather that Asterisk currently always uses 20ms packets for RTP, although I can't find in the source where that's defined, unless it's in chan_zap.c. I'm guessing from [http://lists.digium.com/pipermail/asterisk-users/2003-September/019490.html] that it's not

Hey, all. I'm trying to do some fun with auto-provisioning of Polycom phones, and one thing that would make life easier for me would be if I could have a per-phone sip.conf file. If not, no biggie -- but if there's a way to do an include (as per extensions.conf) or something, that would be great. I've gone through docs, and an older version of "Asterisk: the Future of

Greetings fellow listers, I have an application where I have approximately 300 files that I playback individually or in blocks to simulate "text-to-speech" in a "less mechanical" voice than normal Allison files provide. These files are presently in GSM format and sound pretty good when I play them on my computer speakers or on my in-house

Hello, I want to create channel bank in this case: "channel bank" |-----------------------------------------| | FXS,FXO<----->TDMoE<--|---------------------------------->Asterisk |-----------------------------------------| How can it?

Hi All, First of all, please no flames about the footer of this message. It's being inserted into all outgoing email by the server here without asking me personally first once I send out email. Since I've been very interested in setting up an Asterisk PBX + IVR + Voicemail system to support a small business and a foundation for quite some time now, in order to replace their current ISDN

Hello, Is it possible to install Asterisk on Vmware(centos) from source. Is there any difference or disadvantage for this compared to asterisk running on physical machine. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100811/05a14968/attachment.htm

Hi there! I'm currently considering various PBX solutions for our office telephone network, and would very much like to use Asterisk. Currently, my research is incomplete. I have been recommended to use the above cards, but it is unclear from my Googling whether my configuration will work: - 3x Fritz!Card PCI's in one host. - 3x 6 b-channels. - ~20 Budgetone (and some others)

This message is in MIME format. The first part should be readable text, while the remaining parts are likely unreadable without MIME-aware tools. Send mail to mime@docserver.cac.washington.edu for more info. --8323328-465625440-1002298640=:29463 Content-Type: TEXT/PLAIN; CHARSET=US-ASCII Content-ID: <Pine.LNX.4.33.0110051717531.29463@localhost> ---------- Forwarded message ----------

hi, list i want to know what is the best OS for install Asterisk 1.6.2.9, which should work properly on working system. i want to use CentOS5.2 or CentOS 5.4. Which is better and stable? Thanks for your help. -- Thanks for your supporting, have a nice day. Sucan

Thanks Nicolas for your help with my build problems with vmkit. Yes, I was missing ant -- great catch (I'm on a newly installed machine). That fix with this configuration, gets me to the error below. Any thoughts (build tools, other I may be missing that are not checked for)? Are there steps that you suggest I try to gather more information on the problem? Also -- it seems to be building