[asterisk-users] AstLinux 1.2.0 Released
The AstLinux Team has released 1.2.0. All current users are encouraged to upgrade as this release addresses the bash ShellShock bug. New in 1.2.0: * New Linux Kernel 3.2.x * igb ethernet driver for Intel Atom C2000 * Enable AES-NI support * New sip-user-agent firewall plugin * New versions of Asterisk 11 and 1.8 * Bash ShellShock security fixes A full changelog can be viewed in the release pages: http://www.astlinux.org/release/120-asterisk-11121 http://www.astlinux.org/release/120-asterisk-18300 New AstLinux Documentation Topics: SMTP Local Aliases http://doc.astlinux.org/userdoc:tt_smtp_aliases Updated AstLinux Documentation Topics: Firewall Plugins - sip-user-agent http://doc.astlinux.org/userdoc:tt_firewall_plugins#sip-user-agent --The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 1.0.4 Released
The AstLinux Team is happy to announce the release of 1.0.4. New in this release: -- Asterisk 1.4.44 and 1.8.14.1 -- DAHDI, dahdi-linux 2.6.1 and dahdi-tools 2.6.1 -- wanpipe, version bump to 3.5.27 -- rhino, version bump to 0.99.6b2. Support is now enabled again by default. -- libPRI, upstream patch to add layer 2 persistence option to customize the layer 2 behavior on BRI PTMP lines. (Thanks to Michael Keuter) -- PHP version bump to 5.3.14 to address security issues. -- Security fixes for OpenSSL -- miniupnpd added (disabled by default) to support Universal Plug and Play. (Many thanks to David Kerr) -- mtr added. Network diagnostic tool that combines ping and traceroute. -- Updates to the web interface including the addition of a MeetMe tab, firewall enhancements and UPnP support. For the complete changelog and to download the install images go to the following pages: http://www.astlinux.org/release/104-asterisk-18141 http://www.astlinux.org/release/104-asterisk-1444 The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 1.0.2 Release
The AstLinux team is happy to announce the release of version 1.0.2. This release features several security updates. All current users are encouraged to upgrade as soon as possible. Please see the documentation at http://doc.astlinux.org for upgrade details or the official release pages. Updates: Asterisk (1.8.9.2) DAHDI (2.5.0.2) Rhino(0.99.5b1) Wanpipe (3.5.24) The Sangoma BRI/Hybrid cards (A500 + B700) are now supported via DAHDI Security Fixes: PHP(5.3.10) OpenSSL(0.9.8t) New Features: A Test SMTP Mail Relay feature was added to verify msmtp configuration See the change log on either of these release pages for more details http://www.astlinux.org/release/102-asterisk-1892 http://www.astlinux.org/release/102-asterisk-1443 Enjoy, The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 1.01 Released
The AstLinux Team would like to announce the release of 1.0.1. This version is available with either Asterisk 1.4.43 or Asterisk 1.8.8.3. A full changelog and upgrade (or new install) instructions are available on our website. Please follow the upgrade instructions carefully when upgrading from a release prior to 1.0. http://www.astlinux.org Please note that this release includes a change in the way PATA (ide) devices are handled by the kernel. Those devices are now handled by libata which references the drives as /dev/sdx instead of /dev/hdx. As always, please report any issues (and comments) to the AstLinux mailing list on Sourceforge. (link available at the above website). The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 1.0.0 release
The AstLinux Team is happy to announce the release of AstLinux 1.0.0. This release includes significant changes and improvements over past releases. Specific upgrade and new installation instructions are available at: http://www.astlinux.org Some of the highlights include: * Using eglibc instead of uClibc. This allows binary compatibility with add-ons that are provided as binary only (G.729 CODEC, Fax for Asterisk etc). * Newer Kernel which better supports newer hardware * Support for Jabber/Gtalk * Removed mISDN support (the zaphfc DAHDI driver is included for single port ISDN cards) A full changelog is available on the release pages. We provide versions with Asterisk 1.8 and 1.4. Because this is a major version change, there are some special considerations when upgrading. Please read the instructions very carefully to ensure no step is skipped. http://doc.astlinux.org/userdoc:upgrade-0.7 Please report any issues with the release back to the AstLinux mailing list. Enjoy, The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.8 Release
The AstLinux Team would like to announce the immediate availability of the 0.7.8 release. This release includes either Asterisk 1.4.41 or Asterisk 1.8.4. All current users are encouraged to upgrade to this release to take advantage of bug fixes and other updates to Asterisk. Please note that there is a bug in Asterisk 1.8.4 that will prevent Cisco 79xx phones from registering. A full changelog is available at http://www.astlinux.org Current users can upgrade from the web interface or from the commandline. From the CLI: (Asterisk 1.4) upgrade-run-image check http://mirror.astlinux.org/firmware --should report astlinux-0.7.8 upgrade-run-image upgrade http://mirror.astlinux.org/firmware (Asterisk 1.8) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware --should report astlinux-0.7.8 upgrade-run-image upgrade http://mirror.astlinux.org/ast18-firmware -- The AstLinux Team http://www.astlinux.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
On 03/29/2011 07:16 AM, Gilles wrote: On Mon, 28 Mar 2011 08:20:23 -0400, vip killavipki...@gmail.com wrote: Is anyone using asterisk with fail2ban? Sorry for hi-jacking the thread, but I was wondering if there were a lighter alternative that I could run on appliances? Python uses too much RAM, but I need to find a way to ban hackers from trying to connect to Asterisk from the Net. Gilles, One of our developers on the AstLinux team worked out a plugin for Arno's firewall (iptables based) which performs similar to fail2ban, but uses bash. He called it adaptive-ban. You might be able to adapt it for your use, but as it's written, it's integrated with AstLinux. http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/0.7/package/arnofw/adaptive-ban/ Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] doorphone?
On 03/09/2011 02:57 AM, Dan Journo wrote: could anybody suggest a usable doorphone and magnetic door opener hardphone system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. I would recommend using a normal doorphone, and connecting it to a SIP gateway like the PAP2T. Otherwise, you need a network connection directly into the doorphone unit, and some people don't like that because it can give a hacker/burglar, direct access to your internal network. Hope that helps. Dan Journo That's not always true. Some door phones have a remote unit that connects to the network and a local device at the door, giving some better security. I've used the Valcom VIP-172 phones. They are simple and work well. Very good support if you need to call them. http://www.valcom.com/Home_links/sipdoorintercom.htm Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.7 Release
The AstLinux Team would like to announce the immediate availability of the 0.7.7 release. This release includes either Asterisk 1.4.40 or Asterisk 1.8.3. All current users are encouraged to upgrade to this release to take advantage of bug fixes and other updates to Asterisk. PPTP was added as a possible VPN option. A full changelog is available at http://www.astlinux.org Current users can upgrade from the web interface or from the commandline. From the CLI: (Asterisk 1.4) upgrade-run-image check http://mirror.astlinux.org/firmware --should report astlinux-0.7.7 upgrade-run-image upgrade http://mirror.astlinux.org/firmware (Asterisk 1.8) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware --should report astlinux-0.7.7 upgrade-run-image check http://mirror.astlinux.org/ast18-firmware -- The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.6 Released
The AstLinux Team is happy to announce the latest release (0.7.6). There are several security updates as well as feature enhancements/improvements. All current users are encouraged to update. A full changelog is available at: http://www.astlinux.org Both Asterisk 1.4.39.1 and Asterisk 1.8.2.3 are supported on separate firmware images. Current users can upgrade from the web interface or from the commandline. From the CLI: (Asterisk 1.4) upgrade-run-image check http://mirror.astlinux.org/firmware --should report astlinux-0.7.6 upgrade-run-image upgrade http://mirror.astlinux.org/firmware (Asterisk 1.8) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware --should report astlinux-0.7.6 upgrade-run-image check http://mirror.astlinux.org/ast18-firmware -- The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.5 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.5 with options for both Asterisk 1.8.1.1 and Asterisk 1.4.36. More information about the release is available on our website: http://www.astlinux.org/content/astlinux-075-release Direct links to the installation files are available here: http://www.astlinux.org/release/075-asterisk-1811 http://www.astlinux.org/release/075-asterisk-1436 All current users are encouraged to upgrade to one of those releases. A firmware upgrade can be performed from the web interface or from the command line. Command line upgrade: (for Asterisk 1.4) upgrade-run-image check http://mirror.astlinux.org/firmware (should report astlinux-0.7.5) then upgrade-run-image upgrade http://mirror.astlinux.org/firmware or (for Asterisk 1.8) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware (should report astlinux-0.7.5) upgrade-run-image check http://mirror.astlinux.org/ast18-firmware If you are upgrading from an Asterisk 1.4 base to Asterisk 1.8, you will need to manually update any Asterisk related configuration files. Please ask any questions about this release on the AstLinux-user's mailing list. -- The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with MOH - Asterisk 1.4.17
On 12/03/2010 03:30 PM, Doug Lytle wrote: Napoleón Ernesto López Espinoza wrote: We're sorry, your call did not go through. Any clues about this issue? How about some output from your console when it fails? It's would also be advised to use a much more recent version. Asterisk 1.4.17 has many bugs and security issues that have been addressed in newer versions. 1.4.37 is the latest version from the 1.4 branch. It's quite possible that whatever you're trying to fix is already fixed in that newer release. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.4 Release now available
The AstLinux Team is happy to announce the release of AstLinux 0.7.4. This is a dual release which allows you to chose between Asterisk 1.4.36 or 1.8.0. There are several security updates and other improvements. All current AstLinux users should upgrade as soon as feasible. One of the more significant additions includes preliminary IPv6 support. The two releases can be viewed here. http://www.astlinux.org/release/074-asterisk-1436 http://www.astlinux.org/release/074-asterisk-180 A full changelog is available on those pages. Current users can upgrade either from the web interface or via the command line. upgrade-run-image check http://mirror.astlinux.org/firmware (http://mirror.astlinux.org/ast18-firmware for Asterisk 1.8 firmware) The version should be reported as 0.7.4 upgrade-run-image upgrade http://mirror.astlinux.org/firmware The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommended *WRT router to run Asterisk?
On 11/18/2010 07:52 AM, Gilles wrote: On Tue, 16 Nov 2010 09:42:33 -0500, Mark Deneenmden...@gmail.com wrote: Are you saying ADSL as in a generic term for broadband router or do you really mean that the router also acts as a DSL transceiver? Sorry about that. Ideally, the unit should be both an ADSL modem + router, but apparently, most of them are just routers so that the user would have to turn their ADSL router into a modem/bridge and connect the *WRT-moded router. If someone's been running Asterisk on that kind of hardware for SOHO use, what would you recommend? Apparently, those are hardware that come up often in forums: http://wiki.openwrt.org/toh/d-link/dir-825 http://wiki.openwrt.org/toh/buffalo/wzr-hp-g300h http://wiki.openwrt.org/toh/asus/wl500gp http://wiki.openwrt.org/toh/asus/wl600g I never saw the point of spending $100 for something that is so limited. You can spend a little more and get something like an ALIX board that is so much more capable and still fanless/low power. http://www.pcengines.ch/alix.htm The 2d3/2d13 are very nice for the price. If you really want to run on a small router like this, the Netgear WNR3500 is a decent device and can be found for around $90. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk runs at 100% CPU
Patrick, I observed this same behavior on a system a few weeks ago. If Asterisk was not running, the CPU load would be normal. There were no 'failed' attempts in any of the logs. There was a relatively large amount of bandwidth coming from a specific IP address. (I used iftop to determine the offending address). You probably should upgrade to a newer version of Asterisk. 1.4.21 is pretty old and likely has several security holes which were fixed in newer releases. Darrick On 11/17/2010 12:53 AM, Patrick wrote: I also forgot to add that my bandwidth is highly used (mostly out traffic) since I've detected the attack On Wed, Nov 17, 2010 at 06:46, Patrickasterisk-us...@ict-synergy.be wrote: Dear asterisk users, A few weeks ago I've been attacked by a DOS on REGISTER that I've solved with a fail2ban script. Now, since a few hours, I have my asterisk 1.4.21.2 running at 100% CPU again. I've checked the log and it shows nothing related to failed register or whatever. It just tells me that some of my peers are lagged, even with a verbosity of 1 I've made a SIP SHOW CHANNELS and I've a very strange thing, I got between 4000 and 5000 active channels from peer 127.0.0.1. I have no sip phone on localhost. Here is an excerpt of my command Peer User/ANRCall ID Seq (Tx/Rx) Format Hold Last Message 127.0.0.1(None) 38567737700101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 1623666249 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 1478349241 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 1830524844 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 1688182896 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 1391124899 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 2692644729 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 2043438815 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 3226298375 00101/1 0x0 (nothing) No Rx: REGISTER 127.0.0.1(None) 17042946600101/1 0x0 (nothing) No Rx: REGISTER It is not a configuration issue causing loops because my config has not changed since months. Any help is appreciated Best regards, Patrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any good guides for installing Asterisk on Embedded systems like Alix boards?
Bruce, AstLinux supports dhcp and dns as well as several vpn options including openvpn. You can download a live ISO image to test. http://www.astlinux.org Darrick On 11/08/2010 08:34 AM, Bruce B wrote: Thanks for the input. I am looking to use it as a DHCP server as well. And I also I want it as a VPN server so that I can securely log in to it from time to time to monitor it's state. The Alix board with pfSense can nicely do VPN and DHCP (no Asterisk). Wondering if those two service would play nice along with Asterisk. Thanks, On Mon, Nov 8, 2010 at 3:04 AM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.co...@xorcom.com wrote: On Sun, Nov 07, 2010 at 01:10:01PM -0500, Paul Belanger wrote: Most desktop distros are just too bloated for an embedded solution. I use Debian on an Alix system as my home router. It runs Asterisk as well. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] inbound call issue...
You didn't say which version of Asterisk you were using. insecure=very is deprecated in favor of insecure=port,invite Many of the VoIP providers do not have this right in their examples. Darrick On 11/08/2010 05:52 PM, Gregory Malsack wrote: Not sure if you read the configs I attached, but that line is already in there... Guess again... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, November 03, 2010 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] inbound call issue... insecure=very should fix it. On Wed, Nov 3, 2010 at 4:08 AM, Gregory Malsackgmals...@gmellc.com wrote: Can anyone tell me why my inbound calls keep getting rejected with 401? Here’s the debug information: --- SIP read from UDP:147.135.32.221:5060 --- INVITE sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact:sip:4144038...@147.135.32.221:5060 Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: application/sdp v=0 o=2475098871 10 10 IN IP4 147.135.2.247 s=- c=IN IP4 147.135.2.248 t=0 0 m=audio 15502 RTP/AVP 0 18 8 96 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:96 iLBC/8000 a=fmtp:96 mode=30 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 - [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- (11 headers 14 lines) --- [Nov 3 02:08:40] VERBOSE[7207] netsock.c: == Using SIP RTP CoS mark 5 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Sending to 147.135.32.221 : 5060 (NAT) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Using INVITE request as basis request - 31007e...@147.135.32.221 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Found peer 'trunk_1' for '4144038968' from 147.135.32.221:5060 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- Reliably Transmitting (NAT) to 147.135.32.221:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281-;received=147.135.32.221 From: Wi Msip:4144038...@147.135.32.221;user=phone;tag=9bbc To: Gregory Malsacksip:s...@216.26.109.22;tag=as4fffe111 Call-ID: 31007e...@147.135.32.221 CSeq: 1 INVITE Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2dd58be8 Content-Length: 0 [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: Scheduling destruction of SIP dialog '31007e...@147.135.32.221' in 32000 ms (Method: INVITE) [Nov 3 02:08:40] VERBOSE[7207] chan_sip.c: --- SIP read from UDP:147.135.32.221:5060 --- ACK sip:6087294...@216.26.109.22:5060 SIP/2.0 Call-ID: 31007e...@147.135.32.221 CSeq: 1 ACK From: Wi Msip:number f...@147.135.32.221;user=phone;tag=9bbc To: usernamesip:s...@216.26.109.22;tag=as4fffe111 Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Max-Forwards: 70 Content-Length:0 Here’s the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register = 6087294351:sip password@sip.broadvoice.com [trunk_1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=6087294351 secret=sip password username=6087294351 insecure=very context=DID_trunk_1 authname=6087294351 dtmfmode=inband dtmf=inband canreinvite=no [guest] type=friend host=dynamic canreinvite=no context=DID_trunk_1 -- Darrick Hartman DJH Solutions, LLC http
[asterisk-users] AstLinux 0.7.3 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.3. This update contains mostly bug fixes and security updates. All current users of AstLinux are encouraged to update to this release. Updating can be performed from the web interface or from the command line using a few simple commands. For the Changelog and other instructions, please visit: http://www.astlinux.org/release/073 Enjoy, Darrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and QoS
The Astlinux project has been using the HTB queue and a shaper based on Wondershaper for several years. Recently, we ported the work to Arno's Firewall as a plugin. That work would make it usable on generic Linux distribution. To be effective, you need to have traffic classified properly by the applications. You can find Arno's iptables firewall on the following page: http://rocky.eld.leidenuniv.nl/joomla/ Darrick On 07/30/2010 03:06 AM, Jonas Kellens wrote: Hello list, anyone here using Asterisk together with HTB for queing incoming and outgoing packets ? I've tried to subscribe myself to the Mailinglist of the Linux Advanced Routing Traffic Control project, but I get no confirmation. This list seems dead. It seems my test case with HTB is not giving any noticeable results. Can I ask questions on this mailinglist ? Perhaps you can give my other QoS-implementations like MasterShaper, if it works well together with a firewall that uses iptables. Kind regards, Jonas. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Are your sound files being transcoded or played back in their native formats? On 04/21/2010 12:25 PM, bruce bruce wrote: Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On 04/12/2010 08:17 AM, Fred Posner wrote: On Apr 12, 2010, at 9:12 AM, --[ UxBoD ]-- wrote: Perhaps if there was a Asterisk RBL we could all contribute to; for which we could then hook into and drop any connection where a source IP is listed ? -- Thanks, Phil I love the idea of a RBL... count me in for contributing. Especially considering the ridiculous response I received from Amazon. (Basically told me to submit host, destination, port, proto, and log... which of course was already included in the original complaint) I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ We could establish something similar to that for VOIP attacks. It may not be exactly a trivial system to maintain such a list. (removing IP's after X amount of time, disputing false claims etc). Maybe someone could contact spamhaus to create a list for VOIP since they seem to have a nice system in place? Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
On 04/12/2010 12:05 PM, Randy R wrote: On Mon, Apr 12, 2010 at 6:51 PM, Darrick Hartman dhart...@djhsolutions.com wrote: I don't think anyone else brought up the Spamhaus DROP project. It's a blacklist of IP addresses and address ranges which are known to ONLY be used for malicious purposes. http://www.spamhaus.org/drop/ Because this is in Amazon's interest, THEY should set up a way to report these. Once you detect (in a script) that this is in their range, a redirect would feed their own log with all the data they'd need to proceed. This would work well, especially if they made you register your calling IP to them, or authenticate. That way your server and IP is on record and the report authenticated. Isn't this reasonable? Randy, That only addresses EC2 (and assumes that Amazon has any interest in protecting their reputation). What about attacks that come from other locations? Granted it's pretty easy to buy time on an EC2 server so this may be the primary source for a period of time. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] canary_thread
Danny, I haven't been able to test further, but I've seen the same issue since I upgraded to 1.6.2.6. The astcanary does not appear to stay up more than a few seconds when asterisk is initially started. On this same system using 1.6.2.2 (the previous version I was running prior to the upgrade) it was running fine. I never did like birds anyway... Darrick On 04/01/2010 04:22 PM, Danny Nicholas wrote: You’d think that this is/was some kind of April fool message, but it is a real 1.6 warning http://lists.digium.com/pipermail/asterisk-commits/2008-May/022745.html Since 1.6 has more multi-thread capabilities, the good folks at Digium/Asterisk made this warning program to keep runaway threads from crippling Asterisk. When you get this message, the mine is about to collapse (potentially) on your Asterisk instance. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marcus Vinicius *Sent:* Thursday, April 01, 2010 4:06 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] canary_thread People, Anybody knows what mean this message in my CLI: [Apr 1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) mediagw*CLI Asterisk: 1.6.2.6 -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Sometimes you need to look at the cost to pull new wire too, not just the cost of the phones. There are a few cases where the channel banks + analog phones make sense, especially when the analog devices are already there. Sent from my BlackBerry® wireless device from U.S. Cellular -Original Message- From: hin lee hi...@yahoo.com Date: Tue, 30 Mar 2010 08:25:19 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.1 released
The AstLinux Team is happy to announce the release of AstLinux 0.7.1. This is a bugfix release which includes updates to Asterisk (1.4.30), Dahdi and several other items as detailed in the Changelog. http://www.astlinux.org/release/071 Existing users can upgrade from the web interface or from the CLI. From the CLI execute the following: upgrade-run-image check http://mirror.astlinux.org/firmware (should report that Newest available version is astlinux-0.7.1) then do the upgrade: upgrade-run-image upgrade http://mirror.astlinux.org/firmware Reboot After rebooted, you'll need to check two more items: Upgrade firewall plugins: upgrade-arno-firewall check upgrade-arno-firewall upgrade Install sound files: **NOTE sound files are not installed by default starting with 0.7.1** upgrade-asterisk-sounds upgrade core en ulaw Enjoy! Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Smallest possible Asterisk VM
AstLinux is well under that. You could build a custom image that contains only what you want and have it under 30M. We have support for sqlite3, but not mysql or postgresql. You would have to build your own package to include python. Our build environment is based on buildroot, but has been heavily modified to suit our needs. Feel free to ask questions on the astlinux-devel mailing list. http://www.astlinux.org Darrick On 02/02/2010 12:02 AM, Ben Schorr wrote: I think Astlinux comes in under 100MB. Ben M. Schorr -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Frank Church Sent: Monday, February 01, 2010 19:41 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Smallest possible Asterisk VM How small can an Asterisk system be, in terms of disk space utilized? I am looking for just asterisk, with mysql, postgresql, or sqlite, with PHP and Python. After finishing the build and removing the tools, how small can the whole system be? 100Mb, 200Mb? Can packages be used to build the whole system, like using debs and rpms alone? /vfclists -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.7.0 Released
The AstLinux Team would like to announce that the 0.7.0 version of AstLinux is available for download. There have been many significant updates in this release including updating to the latest Asterisk Release (1.4.29), moving to DAHDI (2.2.0.2) along with several other system updates. For a complete list of changes, read the changelog available on the download page: http://www.astlinux.org/release/070 At the same time, we'd like to officially launch our new website(s). We added a separate documentation site http://doc.astlinux.org and updated the main http://www.astlinux.org website to have a more user-friendly layout. Downloads are available directly on that site rather than on the Sourceforge site. A big thank you to everyone who contributed either as a developer or an end-user providing feedback. We will also be releasing official images which contains Asterisk 1.6.2.1. Those release files will be available in the upcoming week. We will continue to support the 'long term stable' release (currently the 1.4.x branch). The AstLinux Team -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] You won't help me anymore?
On 01/10/2010 11:38 PM, hadi motamedi wrote: FWIW, he did post his question yesterday. I've just taken a look and one potential issue I've spotted is that the external server he mentions is 192.168.0.139, which is part of the 192.168.0.0/16 http://192.168.0.0/16 netblock reserved for private networks. So while the server might be 192.168.0.139 on it's own LAN, I suspect that won't be its public IP address. Other than that, I suspect there might be an issue with the dialplan. The OP posted an excerpt from his sip.conf but I suspect we'd need his extensions.conf or extensions.ael (whichever or both he's using) before being able to help further. Thank you very much for your reply . My Asterisk CallerId issue is as the followings : My Asterisk has sip connection with external sip server and sip inbound and outbound calls are ok . But for the sip inbound calls when the external sip server sends SIP INVITE with CallerId field in the range of my Asterisk sip phones the call will be rejected . For example , please imagine that my Asterisk sip phones are at 667 range so when the external sip server places sip inbound call with SIP INVITE CallerId as say 667 2020 the call will be rejected . But if he modifies his CallerId to say 021 667 2020 (i.e. with area code included) the call will get through . Can you please let me know what is the problem here ? It sounds like a dialplan issue where you don't have a pattern which matches 6662020 while you do have something that matches 0216672020. Without seeing the dialplan, we can only guess. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rick Huebner Sent: Monday, December 28, 2009 4:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Looking at Asterisk for 8000sq/ft residential use My brother-in-law is finishing up his McMansion and I've done all of the low voltage wiring and am starting the trimout. We are batting around what to do for a phone system and I'm torn between a Panasonic TAW824/TVA50 and using an Asterisk implementation. I'm very strong on the networking/linux/basic hacking(old school, not criminal) side. I've downloaded the Asterisk VM and have some implentation questions before we make a decision. Of course we are running out of time because I need to order either RJ-11 or RJ-45 keystones for the plates to finish the trim out. We have Cat5e run everywhere so that won't be a limiting factor. Basic info: 8000sq/ft under air, 11,000sq/ft under the roof 17 phone handset outlets 15 phone jacks for potential use behind TVs 2 fax lines 1 alarm line 3 voice POTS lines 1 fax POTS line Pentium 4 old Dell with 1gig RAM to use with Asterisk if selected Requirements 1. Page over all handsets in intercom mode. They have kids and want to be able to yell over the phone if needed to find someone. 2. Easily call from room to room. Speed dial buttons would be ideal. 3. Multiple voice line support for the office phones. 4. Unique ring tones on the phones for internal calls versus external so you can tell by listening if it is inside or outside. 5. If possible, unique ring tones for the various external lines in the offices. I can't believe anyone would use RJ-11 any more. You can multi-purpose RJ-45 jacks to work with POTS lines. Run everything down to a central panel and send pots over the jacks that you need to. That way if you decide you need/want to go IP in the future, you're all set. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dell Server suggestion
On 12/23/2009 03:48 PM, Fred Posner wrote: On Dec 23, 2009, at 4:21 PM, Sascha Ferley wrote: Hi, I am in need of ordering a new server here for our asterisk solution. Since the corporate standard is Dell we need to stick to a dell server. We used to deploy 2900III without any issues, however now they are almost not available any more and are looking at a new solution. Has anyone tried any of the new Dell R (series) servers with Asterisk, utilizing Digium PRI cards? The biggest issue I can see is that in the future we may want to get a transcoder card, however none of the new servers have a standard PCI slot available any more as with the new Nathelem chips having gotten rid of the basic bridge I guess. Any suggestions? Thanks S. Personally, not a big fan of Dell Servers. That being said, I've deployed Asterisk in many of the new R series 1 2 U servers with Digium cards. Initial deployments have been without issue. Longterm have had the random drive failure and heat issues I've come to experience with Dell servers. Fred, To be fair, you see heat issues (and drive failures related to heat issues) in nearly any 1 or 2U server that's not in a properly cooled environment. 1U and 2U servers work well in data centers with 60F cooling. Not so much in your normal office with computers crammed in a closet. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM 400 hardware(?) issue
On 12/21/2009 05:34 PM, Steve Totaro wrote: On Mon, Dec 21, 2009 at 6:06 PM, Greg Woods g...@gregandeva.net mailto:g...@gregandeva.net wrote: I am having to abandon asterisk after having used it for 2.5 years due to this problem. Every couple of days (sometimes more often, sometimes less), the machine will lock up because the TDM board or the Dahdi driver goes south. /var/log/messages starts filling up with repeated messages: kernel: TDM PCI Master abort The card I have is: *CLI dahdi show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 5 OK 0 0 0 I asked about this once before and I am asking again in desperation, as I have had to shut off my asterisk server, take all the VOIP phones out of service, and go back to the bad old days of a single cordless phone base with a couple of handsets and a crappy old WalMart answering machine. This sucks. Last time I asked, the most helpful answer I got was privately, saying that either my card is bad, or I might need a different motherboard (but there is no way I know of to know which motherboards would work and which would not; the last thing I would want is to go through the expense and major hassle of swapping motherboards only to find out that the problem is not fixed). The only decent diagnostic I have is that if I catch it soon enough, before the system totally locks up, then stopping asterisk, restarting dahdi, and starting asterisk gets things working again (until the next incident). Also, I can go into asterisk -r and do dahdi show status and the card doesn't have any alarms; the output is the same as above, even as the PCI Master abort messages are spewing into the syslog. If my Wildcard TDM board is bad, is there anything I can do about it, or am I just S.O.L. after this much time? The blasted card costs as much as a new machine; either way I can't afford it right now. I don't want to abandon asterisk as it has so many nice features, but I am running low on alternatives at the moment. --Greg Greg, Why don't you contact Digium tech support? They should be able to help you narrow down the problem. Cards do go bad from time to time. Now on to Steve's reply... How many lines are you talking about? In light of your budget issues, I would switch to quality SIP provider and have my numbers ported. That would most likely be cheaper in the long and short run, and more reliable depending on the vendor and your internet connection. I agree, especially for a small office or home. You can set up most SIP providers to failover to your cell phone if there is a problem with the SIP connection. Do this and you won't need a hardware card. Other options are going back to old versions of Asterisk. What version are you running? What was wrong with the version from 1.5 years ago? Maybe your card likes being a Zap device, rather than a DAHDI. Seriously? This makes no sense at all. Even early TDM400p cards will work with dahdi, usually better than they did with zaptel, but no worse. The version of Asterisk and zaptel from 1.5 years ago is likely full of bugs that have been fixed by recent versions. Are you still driving your 1978 Ford Pinto? Someone from Digium made a post to that effect a week or so ago. If you're running version 1.4, you should be running 1.4.28 with the latest version of dahdi. You could make a cron job to reboot the machine at midnight, daily. Bandaid/ducttape? The only thing this may 'solve' is a memory leak. It's really hiding the underlying problem. I have a box full of Digium cards with all sorts of modules, I could sell you what you need for the price of postage but I really think SIP is your silver bullet. That would be nice of you, but he should find out the problem before throwing more hardware at the issue. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live CD - do you think they are worth doing?
On 12/20/2009 10:10 AM, jon pounder wrote: Randy R wrote: I might try a live cd once or twice, or use it to boot a dead computer or one that is not mine, BUT for anything with any sort of time investment in settings to try anything you lose it all with a live cd so why bother since if you can't try it all in one session, you have to start over. Live usb sticks are another matter (assuming your bios actually reliably boots them) at least you can save your changes and pickup where you left off the next time. Hi, Curious, do many of you check out software or projects when they have a live CD or does that make any difference to you? Does anyone know if the general public (not reading this kind of list) is attracted to a Live CD more than an Install one? thx, /r AstLinux is available as a live CD. You can set up persistent storage on a hard drive or CF card. It's actually pretty easy using unionfs. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sendmail
On 12/20/2009 11:38 AM, meetmecall wrote: I used msmtp for delivering mail and this is the procedure I documented once, based on info I found on the internet. I hope it is of help. msmtp also has a rudimentary 'queue' option if you use the msmtpQ/msmtpq scripts -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] b option in Directory
On 12/02/2009 08:32 AM, Martin Roy wrote: I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b option that let you enter the first name OR last name of a user. I see that to make this work I need a patch. I'm wondering how can I install this patch as it's an option one of my customer would like to have but I never had to deal with patch before. I usually just take the release version of asterisk and install it as is. P.S. I would like to keep the version 1.4.21 because it's the last version that I know of that use Zaptel by default instead of DAHDI. You do know that you should be able to compile against Zaptel throughout the 1.4.x series. It's worth the effort to upgrade to Dahdi though. Several improvements. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom retrieve call from hold
Mike Diehl wrote: On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote: Mike Diehl wrote: On Friday 27 November 2009 11:09:02 am Noah Miller wrote: Hi Mike - I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and get it back. It goes on hold just fine. But when I press the resume button, nothing happends. Anyone seen this befor? Any ideas on where to start to fix it? Nope, never seen that one, and I've worked with a LOT of Polycoms. Which SIP/bootrom versions? What asterisk version? I'm running sip 1.6.6.0036 and bootrom 3.1.2.0011 and Asterisk 1.4.0-beta2. You are running phone firmware that is many years old and a very old version (not even an actual release) of Asterisk. While it shouldn't stop working if it was working, moving to a more recent (and supportable) version of the phone firmware and Asterisk would be wise. The phone is a Polycom 501; it's been discontinued. I am working on a testing/migration plan to move to the latest Asterisk 1.6.x, but I'm hesitant to upgrade a system that doesn't currently work right. While the Polycom 501 has been discontinued, you should still be using the latest supported firmware for that phone which is 3.1.4 http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html The 3.x releases of the Polycom firmware have big improvements over the 1.6.x version you're currently using. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom retrieve call from hold
Mike Diehl wrote: On Saturday 28 November 2009 04:48:13 pm Darrick Hartman wrote: Mike Diehl wrote: On Saturday 28 November 2009 06:54:16 am Kevin P. Fleming wrote: Mike Diehl wrote: On Friday 27 November 2009 11:09:02 am Noah Miller wrote: Hi Mike - I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and get it back. It goes on hold just fine. But when I press the resume button, nothing happends. Anyone seen this befor? Any ideas on where to start to fix it? Ok, so I've got a very old firmware on my phone that needs to be upgraded. If I remove my provisioning file(s) from my ftp server and rev the firmware, would this completely wipe out any and all prior configuration on the phone? If this would work, I could elliminate about 50K worth of XML from the equation. Then I'd go in and configure the phone via the web. If I have a working phone, then I can conclude that either the upgrade fixed it, or the XML provisioning file I was using was at fault. Does this sound right? Does anyone have a better/different idea? I'd configure the phone using the XML files, but take a look at the method that Karl Fife has documented here: http://www.kfife.com/voip/ Minimal changes are made to files. The base config files are never touched which makes upgrading firmware versions super easy. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem in installing Asterisk 1.4.13 on my Debian 3.1 server
Also install a recent version! 1.4.26.3 would be the latest in the 1.4 release series. Using something as old as 1.4.13 is not recommended. Alex Balashov wrote: You need to install 'gcc' and 'g++' and associated libraries and headers. hadi motamedi wrote: Dear All Please be informed that I need to install Asterisk 1.4.13 on my Debian 3.1 server . But I got the following message when trying for #./configure : error: no acceptable C compiler found in $PATH Can you please do me favor and let me know what is the problem ? Let me thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What happened to netxusa?
No problems resolving anything here. Website appears to be up. Maybe they had a temporary equipment issue. Matt Darnell wrote: On Wed, Nov 11, 2009 at 1:11 PM, Matt Florell astma...@gmail.com wrote: They had a nice booth at Astricon and everything. Haven't heard anything about them going down, this might just be an unfortunate IT management incident. Both their toll free and fax numbers go to a re-order message...seems like the worst. -Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with concurrent VoIP calls
John Timms wrote: Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Most of my installations are Soekris net5501's with 512MB ram and a 500mhz Geode LX processor. Unless Ubuntu is running a ton of extra junk in the background, that processor should be more than adequate. Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the top command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will skip occcasionally, but cell phones have perfect quality. Your only connection to the PSTN is via SIP right? Then this is likely coincidental that 'landline' calls are different than 'cell phone' calls. The ONLY possibility is that the problem is with your SIP termination provider, but even that is unlikely. As Fred pointed out your DSL connection is likely the cause. Do you have any traffic shaping on the network? If not, you really should have a firewall that's capable of prioritizing voice traffic over bulk data traffic. What is the actual down and up speed of your DSL connection? I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Use this calculator to see how much bandwidth 10 concurrent calls will take. http://www.asteriskguru.com/tools/bandwidth_calculator.php Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] programming phones
Ott Rose wrote: I have question thats not really about astrisk but I figure you guys are doing this sort of thing. We use Aastra 6757i phones. there is some support for XML. the question is how would i go about learning to customize these phones? Read the manual on the Aastra website. It's actually quite comprehensive and not really directly related to Asterisk. It would however be interesting to see examples of what other people are doing with the XML on Aastra's or other applications on Polycoms. There was a guy at the Polycom booth during Astricon that had a very cool medical application using the Polycom VVX1500 phones. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent cell phone voice mail capturing call
Russell Horn wrote: Hi, I've a DID number that gets passed to three internal phones and a cell phone via my outbound IAX trunk. If the cell phone is off or out of coverage, its voice mail captures the call. What's the best way to avoid this? Is there a recommended way to force the cell phone user to press 1 before the call is passed there ala google voice? Or is there another way to detect the presence of the answering machine rather than a human? Thanks, Russell. Require the cell phone user to press a button to accept the call (much the same way that the followme app does). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] deploying asterisk
aster...@opensourcesolution.in wrote: hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever call comes to the no it should be routed to phones which ever phone is free. guys plz forgive me if i am not able to make it clear. your support n guidance will be highly appreciated. thx Let's be realistic here. You need to 'drink the koolaid' before you install it for a client. What I'm saying is you really need to install Asterisk for yourself and get a good understanding of how it works before attempting to sell and install it for a client. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mISDN and B410P questions
Oliver, I have some comments, but am looking for a good answer to this as well. 1). 2.6.27 kernels to include a version of mISDN v2.0. It's not as up to date as the full version on the mISDN.org site. 2). From my understanding chan_lcr is the preferred way to connect Asterisk to mISDN v2.0. From what I've read it does require some form of the LCR software to function. It was a whole level of added complexity that I didn't want to add to the AstLinux project (unless we're forced to do so to support mISDN in Asterisk 1.6+). From a project maintainer's point of view, having mISDN functionality in dahdi is a total win, provided that enough European ISDN hardware is supported. A more interesting approach are these new ISDN cards that appear as a network interface to the system (using a Realtek chip I believe). A few of the ISDN users who use AstLinux are starting to look at those cards. What we've decided to do at this point in the AstLinux project is to continue using mISDN v1 with Asterisk 1.4 releases. We have not been successful in building chan_misdn against Asterisk 1.6 (but then again, we haven't tried very hard either since we're hoping for a better solution as mISDN v1 has more than it's share of issues). Darrick Olivier wrote: Hello, I'm evaluating to possibility to use chan_misdn as a short term workaround, in case latest Dahdi is not stable enough for what we are planning to do (we wish to use Junghanns and Digium BRI hardware with Asterisk 1.6) . I've read www.mISDN.org http://www.mISDN.org but still have a couple of questions : 1. Is correct that in a 2.6.27 (and up) enabled kernel, the embedded mISDN version is 2.X ? 2. Is it correct that Asterisk MUST use chan-lcr to access this mISDN software or is it still possible to install mISDN 1.X to be able to use chan_misdn ? 3. Am I correctly understanding README in Dahdi-linux when I think you can switch Digium B410P support from dahdi to chan_misdn, just editing /etc/dahdi/modules file ? 4. Would you trust chan_misdn as a valuable short term solution for ISDN BRI with Asterisk 1.6 ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IMAP voicemail using subfolders fails.
Barry L. Kline wrote: Kevin P. Fleming wrote: It's not present in the current 1.4 doc/imapstorage.txt file, or any later version. I don't even know why the storage format would matter, since that would be very specific to the IMAP server that is managing that folder. Hmmm http://markmail.org/message/up3rfmdk2kjf6r7y is a link that contains the contents of a README file that looks like it came from Digium. About half-way down is: -- Mailbox Format -- Mailboxes should use the mbx mailbox format. The mbox format does not support concurrent access to mailboxes, which can cause deadlock or strange behaviors. You can convert mailboxes from mbox to mbx using mailutil: Perhaps that came from a different product? I think that I'm going to just go ahead and implement IMAP VM and see what happens. Barry, I don't think that Maildir or a database backend solution (such as Exchange) suffers from this same limitation. I would be more interested in knowing how sensitive this would be to latency if using an IMAP server that isn't on the same device as the Asterisk server (or perhaps even a remote IMAP server)? Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon
John Todd wrote: On Oct 17, 2009, at 7:47 PM, Michael Graves wrote: I'm told that they will show up on the event site in about three weeks. On Sun, 18 Oct 2009 02:29:48 + (UTC), Jeff LaCoursiere wrote: Wish I could have made it :( Is there a possibility of a collection of the talks/slides/handouts/videos/presentations for download? Even pay for? Cheers, j The presentations will be available real soon now but the videos may take a bit longer. Indeed, there will be a cross-section of videos available soon. We're working on the schedule for these, but it takes some time to post- process the videos and then we're probably not going to put them up all at once, nor will all of them appear. Three of the four tracks were taped, and we'll pick some highlights (I'm taking suggestions - email me with your ideas if you liked a particular talk or want to see something specific.) We have to balance a few things - if we put all the talks up, there is a fear (not universally held, I might add) that it will effect attendance next year. Even a few percentage points would make the conference go from what is essentially a break-even to in the red and that's something we're trying very, VERY hard to avoid. However, posting the best talks will also excite people about attending next year, so that's a positive for the conference. John, It would be great if you could make more of the talks available to those that attended the conference. I know there were a few times where two interesting talks happened at the same time. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - DECT SIP Phones
Randy R wrote: On Sat, Oct 17, 2009 at 7:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: | I have three Snom M3s at the moment but getting pretty fed up with | the | issues :( I am UK based and would be interested to hear of other | peoples The S685IP has no headset jack AFAIK. If you want to use a headset or you don't need Bluetooth, get a S675IP. They're great, they do g722 wideband and you have plenty of company in the asterisk world to give peer support. We have a bunch of Gigaset owners on our weekly conference and we've even gotten Siemens Gigaset division to make a significant firmware change bexause they're listening to what we users have to say. Panasonic had some nice looking SIP DECT handsets at the Expo for Astricon. Looked to be robust and more business class than some of the other devices. No idea where or if you can get these yet as it sounded like a fairly new product. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best afordable router with QOS for *
John Knight wrote: You know, I'm not entirely sure. I've never thought about using it outside the context of Tomato. Does anyone else know if that's a standalone (and hopefully architecture independent) package? Michelle Dupuis wrote: I like the Qos functionality. Is that a linux based package available for other distros? A few of the developers for the Astlinux project ported some traffic shaping tools into the Arno's IPtables firewall project. It's a set of scripts and config files which make it easy to implement several iptables related functions. The QOS has several features which are geared specifically to optimizing VoIP traffic. http://rocky.eld.leidenuniv.nl/joomla/index.php?option=com_contentview=articleid=45Itemid=63 (perhaps a cleaner link) http://freshmeat.net/projects/iptables-firewall/ Astlinux has used this firewall script for several years. Recent versions include a nice web interface to access many of the features. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Channels got stuck in asterisk 1.4.18.1
das sandesh wrote: Hi All, Today I faced a problem with channels getting stuck. We use asterisk 1.4.18.1, and there were 2 extensions (channels) that got stuck. When I try to do soft hangup channel, it says Requested for soft hangup for that channel, but if we go and check once again those channels are still stuck. Also even after asterisk restart it did'nt go, finally we had to kill the asterisk process and then start asterisk to come back to normal. I wanted to know did any one faced such a problem? Is there any way of getting to know if the channel gets stuck (since in our senario we came to know since the person at the extension(channel) that got stuck was not able to receive calls) or is there a way to eradicate the channel getting stuck? Thank you very much. Regards Sandesh Upgrade to a recent version of Asterisk. 1.4.26.2 is the latest 1.4 release. Not much chance you're going to get help when you're using something as 1.4.18.1. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Aastra - Alert-Info : how to stop auto-answer on call second leg ?
Add this line to your aastra.cfg file sip intercom allow barge in: 0 # don't barge in on existing calls Olivier wrote: Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer. Is there a way to cancel this auto-answer feature on the second leg of a call, either with a SIPRemoveHeader-like application or using something like (before dialing the second leg) : SIPAddHeader(Alert-Info: info=alert-noautoanswer); I've tried many things unsuccessfully such as: SIPAddHeader(Alert-Info: info=alert-community-1);(From an old doc) Best regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hiding voiemailbox/entry from directory
Use the 'hidefromdir' option in voicemail.conf for that particular entry. This should be clearly documented in the example voicemail.conf file. Michelle Dupuis wrote: I have internal mailboxes that I don't want visible to callers going through the directory. Is it possible (in * 1.4) to hide mailboxes fom the directory, without creating a new context? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2-RC1 question
Ira wrote: I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no problems and while everything seems fine i get these message at startup and than all is well. Should I be worried or do i need to let the team know about this? Also, is not finding /dev/dahdi/transcode a problem I should be worried about? And lastly conf2ael always segfaults when I try to run it. it did run once quite a while ago but after I made all changes for the 1.2=1.62 move it stopped working. Thanks, Ira [Sep 6 12:21:55] NOTICE[3045] cdr.c: CDR simple logging enabled. [Sep 6 12:21:55] NOTICE[3045] loader.c: 142 modules will be loaded. [Sep 6 12:21:55] WARNING[3045] translate.c: plc_samples 160 format f [Sep 6 12:21:55] ERROR[3045] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory You're not going to find that device node unless you have a decoder card (the TC400B card from Digium). My guess is you don't have one. I thought there was some discussion about changing that from an 'ERROR' to a WARNING which is what it really is. A more descriptive note saying something like no transcoding hardware found, I'll do it in software or whatever might be more meaningful. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Polycom sip.cfg is not the same as the Asterisk sip.conf file hadi motamedi wrote: Thank you for your reply . Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Regards H.Motamedi On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney) john@compuware.com mailto:john@compuware.com wrote: Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] waitfordialtone patch
Does anyone have a working patch for the following issue on Asterisk 1.4.26 or an earlier version of 1.6 than 1.6.2? It looks like it got committed somewhere after 1.6.1 was branched and is only available natively in Asterisk 1.6.2.x. https://issues.asterisk.org/view.php?id=12382 I have a crappy Cisco IAD that doesn't seem to be consistent in it's operation when the Analog card attempts to place an outbound call. I've inserted several pauses in the dialing string which helps in most cases, but in the few cases with the Cisco is ready immediately, the extra pauses create other issues. This seems like the best work around, but I'm not really ready to try Asterisk 1.6.2.x in a production setting. Thanks, Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.6.7 released
The Astlinux Development Team is happy to announce the release of AstLinux 0.6.7. This release is a security and bugfix release with no new features. All current users of AstLinux are encouraged to upgrade. Current users can upgrade either from the web interface or by issuing the following commands from the CLI. upgrade-run-image check http://mirror.astlinux.org/firmrware (which should report ... Newest available version is: astlinux-0.6.7) then upgrade-run-image upgrade http://mirror.astlinux.org/firmware New users can download the installation images or ISO images from the Sourceforge project site. Note that sourceforge made some changes so not all 0.6.7 versions appear under 'newest files'. You'll need to browse to the bottom of the files page to the 0.6.7 https://sourceforge.net/projects/astlinux/ Release notes are available here: http://www.astlinux.org/releasenotes/0.6.7 Regards, The Astlinux Development Team ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TDM400P in Soekris net5501-70?
I still don't see what you gain by using m0n0wall and a separate Asterisk install. I can't think of one thing that you would need a separate m0n0wall instance to do that AstLinux can't do on it's own. The web interface has become quite completely in the last few releases. Traffic shaping, firewall, vpn support etc. I don't understand how a VM does anything more than complicate an otherwise simple set up. Darrick Brian McEntire wrote: Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium TDM400P in Soekris net5501-70?
Brian McEntire wrote: Darrick - You seem adamant, and I will look deeper into the firewall in Astlinux! :-) Brian, I am one of the developers, so I happen to like what we've done. There have been some huge changes to the web interface and the overall project in the past year or so. http://www.astlinux.org The one thing running monowall in a VM would do for me is (in theory) make it very simple to move my existing, working m0n0wall configuration. I've been running it for a while, it serves a bunch of DHCP clients, does a little NAT, and has 20 or so specific rules for what can talk to what across the LAN, WAN, and DMZ segments of the firewall. If Astlinux can do all that, and I can grok it easily, it might be easier than running m0n0wall inside a VM. The firewall part of Astlinux is Arno's IPtables firewall. The web interface can handle most (if not all) of what you're trying to do. We've exposed a few more options in our svn trunk, but that's undergoing some big changes right now to support dahdi. I'm running an image based on that right now, but it will probably be another week or so before trunk is stable enough for general use. If there's something you need that's not exposed in the web interface, ask and someone on our mailing list can get you going in the right direction. If you have any problems/questions, ask over on our mailing list or in the #astlinux channel on freenode. I suppose the other thing running m0n0wall inside a VM might do is a little extra security. If the firewall is in a VM and the asterisk part is running on the hardware without access to the LAN ports (which are all owned by the VM) then it *might* make the asterisk install a little more secure or less exposed to automated attacks. Not saying this is a high payoff for me, but another potential pro for a VM setup. That could very well be the case, but I highly doubt you're going to like the results of using a net5501 as a virtual machine host. The hardware was never really intended for that purpose. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to enable dial to a...@asterisk.blurb.com
Alex Samad wrote: Hi The subject line says it all how do I enable this style of call. Pointers to the dns setup and asterisk setup would be great or even search words for google, as I am not sure how to search for this type of request. Alex Alex, Here's a good place to start. http://www.voip-info.org/wiki/view/DNS+SRV Then you would need to enable a few things in /etc/asterisk/sip.conf [general] allowguest=yes context=yourdefaultcontext domain=yourdomain.com Then configure the default context in your extensions.conf file to include routing for your calls. There may be a few more steps, but this should get you going down the right road. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using the PBX Directory from a Blackberry
On 07/02/2009 10:14 AM, Tzafrir Cohen wrote: On Thu, Jul 02, 2009 at 09:53:18AM -0500, JR Richardson wrote: Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure is there is actually tones generated when the alpha characters are pressed, it just doesn't work). Anyhow, on the Blackberry, when you hold down the Alt key and press the alpha character, the device sends out the correct digit tone associated with that character, like on a regular phone keypad. Is it a tone? Or the letter itself in SIP / RTP signalling? This is a 'bug' or 'feature' of blackberry phones. The phones switch the keypad to numeric when in a phone call. You need to memorize abc=2, def=3... Sure would be nice if there was an option to send the DTMF for 5 when pressing the alpha key j k or l, but I don't believe this is possible. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and openvpn and sip
Do you have 'canreinvite=no' in your sip.conf entry for this phone? If not, you should. On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote: Hi John, I already have the ccd dir with the iroute (mandatory for routing to pc/phone connected to vpn client). During the last test I could register and make a call but voice disappears after 1, 2 seconds. I'm trying to understand if it is a bandwidth problem. At the moment I have my phone connected to the openvpn client (which is its gateway) but I have to use the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip (192.168.1.12) is not working. I suppose it is a sip protocol problem: I had to change the sip.conf setting nat=yes to make the phone dial and domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). I keep on working on the vpn since it seems so little is missing to have a clear conversation. Let me know if your tests are successfull. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and openvpn and sip
Giorgio, tcpdump and wireshark are your friends. Instead of guessing, capture a call with tcpdump then look at it with wireshark. Darrick On 06/18/2009 08:58 AM, Giorgio Incantalupo wrote: Hi Darrick, I always set canreinvite=no 'cause it gives a lot of problems if set to yes (and the default is). I made a call with rtp debug on and I noticed that normally, on the asterisk CLI, I see one packet sent corresponding to one packet got (made a test with a local call on our production server). On the other server with the vpn, I get a bunch of sent followed by a group of got...there is something in the way the RTP packets are sent/received by Asterisk and maybe it can be correlated to the missing audio. Giorgio Darrick Hartman (lists) wrote: Do you have 'canreinvite=no' in your sip.conf entry for this phone? If not, you should. On 06/18/2009 07:55 AM, Giorgio Incantalupo wrote: Hi John, I already have the ccd dir with the iroute (mandatory for routing to pc/phone connected to vpn client). During the last test I could register and make a call but voice disappears after 1, 2 seconds. I'm trying to understand if it is a bandwidth problem. At the moment I have my phone connected to the openvpn client (which is its gateway) but I have to use the vpn ip (10.0.0.1) to register the phone, the openvpn server local ip (192.168.1.12) is not working. I suppose it is a sip protocol problem: I had to change the sip.conf setting nat=yes to make the phone dial and domain = 10.0.0.1 to make the voice pass (or at least the first 2 seconds). I keep on working on the vpn since it seems so little is missing to have a clear conversation. Let me know if your tests are successfull. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rhino analog cards
Jeff, Contact their tech support. You will need to send the card in for service, but they may be able to repair it. You should look into getting some sort of surge protection on the analog lines if you don't already have something. The surgegate stuff seems to work well. Darrick On 06/10/2009 08:16 AM, Jeff LaCoursiere wrote: Had a fairly horrible lightning storm night before last, and four of eight ports in a 1.4.20 machine stopped answering. In the CLI: budsw*CLI zap show channels Chan Extension Context Language MOH Interpret pseudodefault en default 1from-zaptel en default 2from-zaptel en default 3from-zaptel en default 4from-zaptel en default 5from-zaptel en default 6from-zaptel en default 7from-zaptel en default 8from-zaptel en default budsw*CLI but in dmesg: r4fxo: module license 'unspecified' taints kernel. rcbfx 1: Rhino PCI BAR0 febff000 IOMem mapped at f8b12000 rcbfx 1: Waiting for response from card . rcbfx 1: Firmware Version 1.f rcbfx 1: Firmware File Version is 1.f rcbfx 1: Hardware version 11 rcbfx 1: G168 07 04 DSP Loader file size = 170 App file size = 48414 rcbfx 1: G168 DSP Ping DSP Version 106 rcbfx 1: G168 DSP Active and Servicing 4 Channels - f rcbfx 1: Starting DMA rcbfx 1: Spotted a Rhino: Rhino RCB4FXO (4 channels) rcbfx 2: Rhino PCI BAR0 febfe000 IOMem mapped at f8b14000 rcbfx 2: Waiting for response from card . rcbfx 2: Firmware Version 1.f rcbfx 2: Firmware File Version is 1.f rcbfx 2: Hardware version 11 rcbfx 2: G168 DSP App Loader Failed 4 rcbfx 2: Unable to intialize G168 DSP rcbfx 2: Starting DMA rcbfx 2: Spotted a Rhino: Rhino RCB4FXO (4 channels) So it seems the second card is fried? A reboot seems to result in the same messages. Trying to arrange for a power cycle - the site is remote. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones dropping registration, but asterisk thinks phones are still registered
There were some serious issues with some of the earlier 1.4.x Asterisk releases. You say it's a production server and can't upgrade because of that. That is the one reason why you should upgrade. There are security risks with certain versions and some serious bugs that were fixed. While I can't say that the problem with go away with an upgrade, you'll get better support if you are running a more recent version. On 06/04/2009 01:08 PM, James Lamanna wrote: Hi, I have a serious problem with Asterisk 1.4.18. Every so often, usually after Asterisk has been running for a few days consistently, phones start dropping registrations. However, when this happens, doing a sip show peer on those extensions shows them as OK. Therefore, I have no way to tell this problem is happening until customers start calling. The only way to fix it is to completely restart Asterisk. Has anyone experienced this? This is a serious problem. I've poured over the logs while and after this happens and there is nothing in the logs that would suggest there is a problem. This is a production server, so I can't just upgrade Asterisk to the latest 1.4 version. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone recommendation
On 06/03/2009 11:47 AM, Jeff LaCoursiere wrote: On Thu, 4 Jun 2009, Rob Hillis wrote: Jeff LaCoursiere wrote: We are still talking about a $175 phone. How about the Polycom IP 320? $85 at 888voipstore. Can't go wrong with Polycom for voice quality. True, Polycom's are brilliant for voice quality, but unlike the Snom, a Polycom /will/ reboot on the drop of a hat /and/ take a damned long time to do it (~45-60 seconds) In addition, the web interface should be taken away and shot - the only real way to configure them is through (T)FTP. They are however, extraordinarily configurable through the XML config and they are very stable. Once they're configured they work very nicely. The lack of a decent number of BLF keys (even with a very expensive sidecar you only get two more keys than a standalone Snom320) puts me off a little. However, for a conference phone, the Polycom's can't be easily beaten. Their handsfree call quality is in a league of it's own. Mainly I suggest it because the OP asked for an inexpensive quality phone. I agree on the provisioning - the web interface is useless, and unless you know how to setup the XML files properly you are doomed to a very frustrating experience. The Polycom 320/330's are nice little phones for the price. There are several resources for configuring the phones from the XML config files. If the config files are sane, the phones don't take that long to reboot. This is probably one of the better examples: http://www.kfife.com/voip/ Karl did a good job commenting in the config files where he made changes. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: Hi, It's a 2mb dedicated leased fibre line, with50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: Hi All, I've a 1.4.15 A*k server supporting several users (approx 80 total, but10 sim calls usually). I've one user who complains of intermittent bad calls, though I suspect the bad calls are across the board, but intermittent. Inbound calls are via in IAX trunk from Gradwell. CPU stats say that Asterisk never uses more than 4-5% cpu, systems idle besides that. Memory seems ok too. Network utilisation is 300kbps. The voice network (clients + server) sit on their own dedicated 100Mb switches. Stats from the switch say its lightly loaded. I've turned on voicefile recording. What we hear, when there is a bad call, is stuttered speech, from BOTH sides (so local SIP client, and remote IAX inbound call). Debug from asterisk just shows the call inbound, answered and then hung up as per normal. I'm at a loss of how to debug the voice issue further, without putting a wireshark PC on the switch, port-mirroring the server and then capturing all of the traffic in a round-robin-type capture and even then I'm not sure what that will achieve. I'm going to switch from IAX to SIP for the inbound calls for that user and see if that helps. Any ideas welcome, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astlinux 0.6.6 Release
The AstLinux project is happy to announce the latest release of AstLinux. Astlinux-0.6.6 is now available for download or upgrade. New users should go to the http://www.astlinux.org website and follow the instructions for a new installation. Current users can upgrade their existing 0.6.x installations either from the web interface or by issuing the following command from the CLI. upgrade-run-image check http://mirror.astlinux.org/firmware (verify that it says the latest available is 0.6.6) then run upgrade-run-image upgrade http://mirror.astlinux.org/firmware The only change from the previous release is the version of Asterisk (now 1.4.25). Enjoy -- The AstLinux Team http://www.astlinux.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.25 and zapata.conf
Asterisk 1.4.25 does work with Zaptel. On 05/31/2009 07:46 PM, bilal ghayyad wrote: Hi All; I discovered that Asterisk 1.4.25 does no thave zapata.conf, any advise? Does it mean that Asterisk 1.4.25 no more support for zaptel and it works only with dahdi? So, what is the latest Asterisk version that is working with zaptel? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Productivity Suite
On 05/21/2009 09:11 AM, Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Robin Rodriguez wrote: still rather frustrating getting the EFK working. If needed I could post that portion of sip.cfg to get you started. Please do! Just having the example could be helpful for those of us preparing to tackle this kind of project. I've remapped the 'line 2' button on most of the Polycom 320/330's to a one-touch messages button. It's basically a speed dial set to dial 8500. The documentation could be better, but at least there's something. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compact, fanless appliance?
Gordon Henderson wrote: On Wed, 6 May 2009, Alan Lord (News) wrote: On 06/05/09 08:28, Gordon Henderson wrote: snip / One little tip: You need to compile Asterisk for an i586 processor as the VIA processor is missing a few (mmx, etc.) instructions that a full blown i686 has. Hi Gordon, I'm using a VIA C7 on a Jetway board (http://linitx.com/viewproduct.php?prodid=11212). This is my cpuinfo. Isn't that an i686 class? Hm. You know what - it's possible my information is a little out of date now.. (quite possibly the wiki too!) I use these processors too, but my test boxes (~6 years old) have the VIA C3 chips in them: Older C3's lacked some of the features, but I believe they all had MMX. Early ones lacked SSE and some other instructions and were best classified as i586. The C7's are definitely i686 compatible. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who has the clever Polycom upgrade system?
Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I remember someone wrote a great document concerning Polycom server provisioning that provided a way to ensure that updates to the firmware did not overwrite customizations. I'll be damned if I can remember where I saw it. It may have been discussed during a VUC session or may have been on this list. Either way, I'm unable to google my way to it. Can anyone point me in the right direction? That would be Karl Fife, of the famous Karl Fife experience. http://kfife.com/voip/ Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk routine maintenance activities
Rob Hillis wrote: Kurian Thayil wrote: On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote: Daily Asterisk restart Do you think its mandatory in production env? Daily? No. However, after implementing a weekly restart of Asterisk, I've found the instance of lockups and CPU utilisation spikes have decreased significantly. Unless you're using some unstable modules, there really should be no need to restart Asterisk. Is there a certain activity that is causing these lockups? I have low power systems which haven't had Asterisk restarted in months many times. Granted, these are mostly low call volume systems, but unless there is a memory leak, you should not needed to restart the Asterisk process. (my guess is one of the modules you are using has some sort of problem). Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk routine maintenance activities
Rob Hillis wrote: Darrick Hartman wrote: Rob Hillis wrote: Daily? No. However, after implementing a weekly restart of Asterisk, I've found the instance of lockups and CPU utilisation spikes have decreased significantly. Unless you're using some unstable modules, there really should be no need to restart Asterisk. Is there a certain activity that is causing these lockups? I have low power systems which haven't had Asterisk restarted in months many times. Granted, these are mostly low call volume systems, but unless there is a memory leak, you should not needed to restart the Asterisk process. (my guess is one of the modules you are using has some sort of problem). This particular system isn't low power - it's a full blown server. Since I don't work at this place, I don't know what people are doing at the time the system freezes up. It's been some time since I updated Asterisk at this site, so they're probably running version 1.4.17 - 1.4.20 there. (it's a voluntary organisation where I've since become sick of (a) the politics and (b) their expectation that I drop what I'm doing to help them, regardless of whether I'm at work or not) Ah. That's probably the issue. There were some significant bugs in some of the releases in that range. If I were to do things again, I'd be running Astlinux on a net 5501 with an integrated hard drive (for voicemail/IVR and so on) Only time I've ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's time to upgrade Astlinux. That's what we like to hear! Did you update to the latest version (0.6.5)? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astlinux 0.6.5 upgrade released
The latest version of Astlinux has been released today. Changes include: Asterisk 1.4.24.1 (security fix AST-2009-03) Asterisk-gui svn 4739 Astlinux web interface 1.4.07 Update Zoneinfo to '2009e' Additional checking is now done when performing an upgrade of the OS. Existing users of Astlinux (0.6.x) can upgrade their system from the web interface or by issuing the following commands from the command line: upgrade-run-image check http://mirror.astlinux.org/firmware (should report something like this) Current version is: astlinux-0.6.4, Newest available version is: astlinux-0.6.5 upgrade-run-image upgrade http://mirror.astlinux.org/firmware The full install images will be available on Sourceforge as soon as my admin permissions are restored for the project (grr sourceforge grrr). In the meantime, new users should install the latest image from Sourceforge, then upgrade to 0.6.5 by performing the previously mentioned procedure. http://www.astlinux.org/ http://sourceforge.net/project/showfiles.php?group_id=170462 Regards, The Astlinux Team ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Practice Advice?
Gabriel - IP Guys wrote: Dear All, I have a asterisk setup that is currently running on version 1.4.15 – I wish to upgrade or migrate this instance to the current asterisk stable, 1.6.0.6. It is my intention to build a FC8 box, then install asterisk from source, and begin to migrate over the configuration. The thing is, this sounds so simple in my head, and I’ve had enough issues with asterisk, to know that life isn’t simple! What I plan to do, is to copy the old configuration over to a box running FC8 – and then compile and run asterisk 1.4.15 – and gradually upgrade it, until I reach 1.6.0.6 – Any input on this matter will be appreciated. Thank you You might reconsider Fedora8. It's support life is gotta be nearing the end. http://fedoraproject.org/wiki/LifeCycle CentOS is generally a better choice for something you want to have in service for any length of time. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inexpensive device for bandwidth management
Mike wrote: I'm looking for a good network device that does bandwidth management. It can be integrated in a router or stand-alone, but must be SIP-friendly. I`ve tried the DIR-655 (latest firmware is SIP-hostile, and the latest hardware revisions can't downgrade to the version that worked well) and the DI-724GU (SIP-friendly, but bandwidth management is automated and not configurable enough for my taste), both from D-link. What else is out there and allows me to do upstream QoS on cable/DSL links? Both D-Link routers were under 200$ (99$ and 159$ respectively) and were perfect price-wise for my target customers. Mike, You could use something like a PC Engine's ALIX board with monowall or pf-sense. The ALIX.2D3 board is around $140. Netgate has a kit with the board, case, ps and a 512MB CF card for $187. AstLinux also has traffic shaping and several other networking features. We include Openvpn, racoon (ipsec), and stunnel vpns, iftop, tcpdump and several others. If you want point and click pfsense or monowall would probably be best. If you're familiar with Linux, you might get more functionality out of AstLinux. http://www.netgate.com/product_info.php?cPath=60products_id=492 http://m0n0.ch http://www.pfsense.com/ http://www.astlinux.org Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Advice
Roland Roland wrote: Hi all, a few month ago I got the task of setting up asterisk for my company. I had 94 employee to set this up for ... I never heard of asterisk before to b honest, so after researching a bit.. I started with a digium card with ZAP though that didn’t work out as the card were flawed.. so ended up setting up SIP for everyone using a SIP callcentric accounts as well as sipura for pstn lines.. now it's working at it's minimal state.. but as am out of the heat of pressure from management.. so now It's time to learn about asterisk the right way as I had lots of help from this mailing list as well as the IRC channel that I'm not sure I could do it again on my own.. so not to add more to my email, I'm seeking advice about the proper way to learn about asterisk from A to Z if possible... How about the Asterisk book? Asterisk: TFOT http://www.asteriskdocs.org/ It's about the most comprehensive guide out there for someone starting out. However, if you really have 94 employees relying on this system, you might consider getting someone locally to consult with you and configure the initial system in a proper way rather than having the boss breathing down your back when it's broke and you can't fix it. The last thing you want to do is set up something that works marginally well and have the company pull the plug on Asterisk because you (admittedly) may not know the best method. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live Support function?
The openfire project has this functionality as part of their package. Requires a Tomcat install, but it works. I set it up on my website as an example, but haven't used it much. (It does work nicely though). Don't see what this has to do with Asterisk though. Darrick Dean Collins wrote: http://openwebim.org http://openwebim.org/ anyone using this one (was just emailed it from another channel) – should have waited more than 5 mins before posting twice. Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net mailto:d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). *From:* Dean Collins *Sent:* Friday, April 03, 2009 7:27 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Live Support function? Hi guys, I’d like to add a LIVE SUPPORT function to my website. Basically I want a client on my desktop that pops up when someone request help BUT doesn’t appear or says offline when I’m not available or have logged out of this function. When a person visiting my website has a question they hot the button to cause a text popup chat to occur. Anyone know of an open source solution? I know there are plenty of commercial hosted options available for a monthly fee but seems like such a simple requirement that something has to be available (especially as I’m only looking for one support client and no need to round robin or multiple agent support or agent cut and paste functions etc). Just need basic text chat function – the terms I’m googling don’t seem to be bringing anything up. (needs to be linux on the server end) Regards, Dean Collins Cognation Inc d...@cognation.net mailto:d...@cognation.net+1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstLinux 0.6.4 available for upgrade
The AstLinux Team is happy to announce that AstLinux 0.6.4 is available. All users of AstLinux are encouraged to upgrade since this release fixes the recently reported security vulnerability in Asterisk 1.4.23.1 Right now a mix up on the Sourceforge site is preventing us from uploading full install versions, but current users of 0.6.2 or 0.6.3 can upgrade to 0.6.4 by using either the 'upgrade-run-image' script from the command line or the upgrade firmware option in the web interface. New versions of the full install will be available as soon as possible on the Sourceforge site. Changes: Asterisk 1.4.24 is included which fixes several bugs and at least one security issue Asterisk-gui was updated to svn 4618 netsnmp was updated to 5.3.2.3 The web interface was upgraded to add several features/improvements An arno-upgrade-firewall script was added to break this away from an init change. This won't really affect users of 0.6.x until they move to 0.7.x which uses a newer version of Arno's firewall. When the time comes, we'll explain the importance. A serial number file was added to trace the version of the firewall config files. To upgrade from the command line: 1). upgrade-run-image check http://mirror.astlinux.org/firmware 2). upgrade-run-image upgrade http://mirror.astlinux.org/firmware 3). reboot as instructed To upgrade from the web interface: 1). Navigate to the system tab on the web interface 2). Select check for new, select the confirm box, and click the Firmware button. 3). Select upgrade with new, select the confirm box, and click the Firmware button. 4). Reboot by checking that confirm button and clicking Reboot. Enjoy The AstLinux Team ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Parked Calls in 1.4.23.1
Leif Madsen wrote: Darrick Hartman wrote: I know the call parking feature changed in 1.4.23.1 to fix some serious issues. I'm seeing a major change though which I find disturbing. A person parks a call by transferring it to the parking position (700). When the timeout value is reached, the call is NOT returned to that device, but rather the 's' extension of the phone's registered context (in this case [unrestricted]). -- Executing [...@unrestricted:1] Park(SIP/100-08217d38, ) in new stack == Parked SIP/100-08217d38 on 7...@parkedcalls. Will timeout back to extension [unrestricted] s, 1 in 60 seconds -- Added extension '701' priority 1 to parkedcalls -- SIP/100-08217d38 Playing 'digits/7' (language 'en') -- SIP/100-08217d38 Playing 'digits/0' (language 'en') -- SIP/100-08217d38 Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/100-08217d38 Is there any way to have the call returned to the device that parked the call (without creating a separate context for each device). For now I created an extension in the [unrestricted] context which sends the call back to the IVR menu, but that's just annoying to the person who placed the call. It should come back to the original device that parked the call. Always did before. There will be a couple of new release candidates going out tomorrow morning. Can you test them once they are announced to determine if this is still an issue? If so, then I would suggest you verify the bug does not exist on the bug tracker already, and if it doesn't, then you can open a new issue. This was probably a miss-use of the system by me as explained by someone else off-list, but perhaps it's still a bug. It ONLY happens with blind transfers. If you use an attended transfer, the behavior works as expected. This is a change from past behavior, but it makes more sense to use an attended transfer to the parking location OR the one step park feature instead. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Parked Calls in 1.4.23.1
I know the call parking feature changed in 1.4.23.1 to fix some serious issues. I'm seeing a major change though which I find disturbing. A person parks a call by transferring it to the parking position (700). When the timeout value is reached, the call is NOT returned to that device, but rather the 's' extension of the phone's registered context (in this case [unrestricted]). -- Executing [...@unrestricted:1] Park(SIP/100-08217d38, ) in new stack == Parked SIP/100-08217d38 on 7...@parkedcalls. Will timeout back to extension [unrestricted] s, 1 in 60 seconds -- Added extension '701' priority 1 to parkedcalls -- SIP/100-08217d38 Playing 'digits/7' (language 'en') -- SIP/100-08217d38 Playing 'digits/0' (language 'en') -- SIP/100-08217d38 Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/100-08217d38 Is there any way to have the call returned to the device that parked the call (without creating a separate context for each device). For now I created an extension in the [unrestricted] context which sends the call back to the IVR menu, but that's just annoying to the person who placed the call. It should come back to the original device that parked the call. Always did before. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pci cards VS patton
On another topic, I would say those gateway are not so easy to configure : - a web server is embeded but it is not documented anywhere and it's GUI is far from natural, - alternatively, you can edit a config file for which a huge doc is available but, as this boxes are not specifically designed to work with Asterisk, doc is not easy to understand (it took me 3 days to find how to register a gateway to an Asterisk server). Did you ever try calling or emailing Patton? I've found their tech support to be very good. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Astlinux 0.6.3 Released
We are proud to release Astlinux 0.6.3. All users of AstLinux should upgrade to this release. Files are available for download at the Astlinux SourceForge project page. https://sourceforge.net/project/showfiles.php?group_id=170462 Updates include new versions of Asterisk, Asterisk-gui, driver updates for wanpipe and Rhino and several updates on the underlying packages. The web interface continues to evolve with a large number of new features including the ability to upgrade to new releases from the web interface. Please see the Documentation section at http://www.astlinux.org for more information about how to install or upgrade from previous 0.6.x versions. Regards, The Astlinux Development Team ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intel Vs AMD
Doug wrote: At 12:33 2/22/2009, michel freiha wrote: Hi all, I took my decision to use Asterisk server for handling my VOIP calls...My next step is to choose the best hardware that I should use i order to have the best performance...Here I faced 2 choices for my hardware (CPU)... 1- Using Intel CPU or AMD 2- Use 32 or 64 bits Can you help me please to choose between the above choices and what is the advantage and disadvantage of each of choices LESS FILLING! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
Jeff LaCoursiere wrote: On Thu, 12 Feb 2009, asterisk_h...@iwishi.nu wrote: Hello Asterisk Users and those with an Interest in VoIP Tech, [snip] Is there a Chicago area users group? If not is there any interest in creating one? We have a group in Milwaukee that meets monthly before the MLUG group. Right now the group is not very active, but we'd welcome visitors from the south, even people from IL. http://www.sewaug.org/ Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Quiet 24 port POE gig switch
OCG Technical Support wrote: A little off topic but I need to put a 24 port Gig PoE switch into a small office – no computer room / rack etc. All CAT5 terminates near the owners desk (smart huh?). I want to put a PoE switch in place, with 24 ports and Gig speed. Everyone I’ve researched so far is LOUD... Chances of finding a PoE switch that is quiet out of the box is about as good as finding a government 'worker'. It's kind of an oxymoron. Of the switches I've used, the Linksys/Cisco line was the loudest. Dlink's were quieter, but still not something you'd want sitting next to a desk. About the only fanless PoE switches I've seen are the smaller Netgear's, but they are not Gigabit. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Interesting observation
I have an interesting observation which I thought I'd pass along to save other people from spending time trying to 'fix' it. One of my clients uses Charter's so called business phone service. They provide 'analog' phone lines over IP. In general, they've worked OK. End users were saying that the phone are cutting out at times. What I've observed is they actually do cut out (meaning all inbound audio is momentarily lost) if a loud noise is created on the local end. This client has a machine shop so you can imagine that at times it does get quite loud. I spent a few hours trying to different setting in the Polycom phones, but finally thought I'd try plugging an analog headset into the Charter CPE device directly. The same behavior was experienced. It appears they have a 'feature' which cuts out the incoming audio if a loud noise (simulated by blowing into the receiver) is experienced outgoing. Pretty much going to a true analog service is the only solution that I can think of. Would be interested if anyone has other thoughts. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Only 8 messages from Asterisk-users To day?
On Tue, 30 Dec 2008 16:51:55 -0500, Doug Lytle supp...@drdos.info wrote: That can't be correct. Doug Could be. It's been quiet all around with the holidays. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone recommendation
Hi Folks, Had a quick search through the archives for softphones and cannot see any recommended ones. My question is what recommended free softphones are out there that can be used with Asterix? I don't really know how many are out there. Is anyone currently using a softphone with Asterix and if so which one and how do you find it? I'm only interested in ones that I can download and use for free. Not interested in any commercial ones that require licenses. Zopier works well and supports both IAX and SIP. Works on Windows, Mac and Linux. http://www.zoiper.com/ Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom low volume
Actually, it could be within Asterisk, but only if you have Zaptel hardware. If you are only using SIP devices, then the problem is with the phone configuration. You really don't provide enough information to determine what is causing your problem. How are you provisioning the phones? What version of the SIP firmware is used on the phones? Are you calling from one phone to the other? Darrick Michael Graves wrote: Probably has nothing to do with Asterisk. You can set the volume and persistence in the phones config files. Michael On Fri, 14 Nov 2008 22:43:45 -0800 (PST), hin lee wrote: Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some progress, anyway...
Philip Prindeville wrote: Just saw from build 2036: Now, to get the following packages to build: misdn asterisk-chanmisdn nistnet rhino strace rp-pppoe Whoops. I'm sure Philip thought he was sending this to a different mailing list. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk installation
Time for you to discover who's your dahdi... Asterisk 1.6 used dahdi and not zaptel. --Original Message-- From: Christian Sender: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk installation Sent: Oct 31, 2008 5:25 PM Hi all, I've just installed the latest v1.6 release of Asterisk. First, I isntalled libpri. Then i installed zaptel with make config at the end of the isntallation as I usually do. Then I installed Asterisk. However, there is no zapata.conf file in /etc/asterisk. I isntalled the sample configuration files. Any tips? Many thanks, Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from my BlackBerry® wireless device from U.S. Cellular ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] network design philosophy and practice
David Gibbons wrote: Two separate networks? Did I miss something? I feel like I'm taking crazy pills! Two separate physical networks means twice the hassle, twice the maintenance, twice the cost, twice the headache. Not to mention the fact that the whole idea of VOIP is to simplify IT and focus on converging data and voice networks. This is what VLANs and QOS do best. I dare say it's what they were designed foe. I can't think of any reason that I would ever recommend two ports per desk to support telephony -- ever. It's ludicrous to think that two ports will be better than one if we're setting up our VLANs and QOS properly. A phone takes very, very little bandwidth away from the desktop and a decent one will support tagging its frames for the alternate voice VLAN. EVER? What about Gigabit networks with 10/100 phones? While some Gigabit phones are available, gigabit POE switches are not cheap, while non-POE gigabit switches are pretty cheap and most business class desktops these days come with gigabit network connections. In a new wiring install I almost always insist on two jacks per location rather than relying on pass-thru connectors on phones. Try giving a few users gigabit access to an Exchange server, then taking it away. They will certainly not be happy! Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey
If you buy your phone from a reputable place they will be able to provide the firmware. --Original Message-- From: Andrew Joakimsen Sender: To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey Sent: Oct 26, 2008 5:45 PM On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote: The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? Which would be great, if Polycom weren't the Firmware-Nazis that they are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sent from my BlackBerry® wireless device from U.S. Cellular ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA for large networks
Vieri wrote: --- On Mon, 9/29/08, Sam Tam [EMAIL PROTECTED] wrote: Why not swap it all with just IP phone? That's because we have almost 400 analog phones already wired in our building (which is very large). So we need to take advantage of the wiring. Also, if we were to convert to an all-IP phone system (non-ATA), we would need to buy more ethernet switches (currently they're all full) and tunnel cables thtough ceilings and walls. In other words, it would cost a lot more than to simply buy ATAs. What I'm looking for however are STABLE, RELIABLE ATAs... Thanks for the feedback, Vieri You'll probably want to use FXS channel banks rather than an ATA. At that kind of scale, I'd call Rhino or Xorcom and have them make the recommendation. You will still end up with a large number of devices and likely several asterisk servers to coordinate all of this. When you really look at the numbers, finding a way to use IP phones may not be that much more than the overall hardware cost involved to do this right with analog lines. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI and verbosity level
Olivier wrote: Hi, Whenever I'm logging in with asterisk -r command, I can see that the verbosity and debug levels are set to a value which is different from the last ones I left when I logged off from CLI. Where are those default levels defined ? I can't see any related option in logger.conf. Any hint ? The verbosity will be at least as high as the last time you entered the CLI. For example if two times ago, you entered with 5 v's then entered the last time with 1 v, you will still be at 5 v's. You can change this behavior using: CLIcore set verbose X where X is the new level you want (2, 3 ...) Hope that makes sense. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on hold for sub tenants
...since everyone else top posted. Take a look at the application setmusiconhold. CLI core show application SetMusicOnHold You can use this in a dialplan as follows: [tenant1incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tenant1sounds/welcome) exten = s,n,SetMusicOnHold(tenant1) [tenant2incoming] exten = s,1,Wait(1) exten = s,n,Answer() exten = s,n,Background(tentant2sounds/welcome) exten = s,n,SetMusicOnHold(tenant2) Use that with the previously supplied info. Darrick carl Lougher wrote: Hi, I tried this but it still uses the default moh. Is there some way to define it based on a context in the sip.conf or extensions.conf??? Taff... --- On Fri, 26/9/08, Nhadie [EMAIL PROTECTED] wrote: From: Nhadie [EMAIL PROTECTED] Subject: Re: [asterisk-users] Music on hold for sub tenants To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, 26 September, 2008, 4:10 AM Hi, i think you can define it like this: [moh-company-a] mode=files directory=/var/lib/asterisk/moh/companya [moh-company-b] mode=files directory=/var/lib/asterisk/moh/companyb regards, nhadie carl Lougher wrote: Howdy, Is there a way to apply a music on hold class to different context user groups? I have multiple clients on my asterisk server and they each want different music on hold. Company A Company B ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon people please post the announcement
Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Lots of questions about this one. There's definitely a demand for it so I can see why Digium would be interested in exploring this option. Time will tell how well it will work. I'm personally not too excited about bolt-on binaries which are probably not compatible with uClibc (and therefore Astlinux). That leaves us in the same place as we are with codec_g729. We're at the mercy of whoever creates these binaries to produce one that will work for us. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Network Monitoring
Dean, I'm using Zabbix to monitor network interfaces, storage, cpu load and a few other things on several asterisk boxes. I'm just looking at adding Asterisk specific monitoring. Simple things like sip registration is pretty easy. Getting the actual status of zap-daddy hardware might be a little trickier. When I get something together I can pass it along. Darrick Dean Collins wrote: Has anyone ever 'released' an Asterisk module that is easily shared/downloadable? Or doesn't the nagios open source code work like that? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Tuesday, 9 September 2008 9:29 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk and Network Monitoring On 14:50, Tue 09 Sep 08, Jacobus van Niekerk wrote: Dear Asterisk Users I'm looking for a solution that can be used to monitor Asterisk and the Telco lines aswell as the network (Servers, WAN LAN links, Router Switches) We use nagios for that. -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com http://www.djhsolutions.com/wiki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
Michael Graves wrote: On Thu, 4 Sep 2008 08:48:47 -0500, Russell Bryant wrote: Asterisk should work fine with any phone that supports that codec. Personally, I have only used it with Polycom phones. Also, again, Asterisk 1.4 only supports G722 passthrough, where as Asterisk 1.6 has full support. Any plans to implement G.722.1 now that it's under a royalty free license? Michael, Royalty free does not mean free. I believe there still is an upfront cost that Polycom is charging. Perhaps Digium can work out some sort of a deal now that Polycom recognizes Asterisk as a valid platform. I'd definitely love to see it supported. It's a great way to actually show some improvement in voice quality compared to the 100 year old copper technology that's in use today. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922
Jay R. Ashworth wrote: On Wed, Aug 13, 2008 at 11:01:46PM -0500, Darrick Hartman wrote: You can get an adapter for the Plantronics that will plug into the 2.5mm jack on the phone. I need the opposite adapter: to plug a 2.5 headset into an RJ-9 Polycom. Anyone know where I can find that? If you use the Plantronics headsets, many of them have a quick disconnect plug part way down the cord. That was what I was referring to, not an adapter which converts RJ-9 to 2.5mm. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slighly OT?.. headset for Linksys SPA922
Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Paul Hales wrote: That's a good question - the plantronics are available with interchangeable ends - which makes them easy to move between different phones. Problem is, the headset button only works for the minijack thingy so if you use the plantronics (plugged into the handset thing) you still need to take the phone off the hook. Unlike the snom 360 where there is a separate socket. - -- Kind Regards, Matt Riddell Director Matt, You can get an adapter for the Plantronics that will plug into the 2.5mm jack on the phone. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
