Re: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform)
Google for some of the How Tos built around Elastix and Trixbox. Both of these are CentOS based as well. good luck. - Original Message - From: James Hankins j...@allpointsmediaworks.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 2, 2009 1:58:15 PM GMT -05:00 US/Canada Eastern Subject: [asterisk-users] Asterisk HA Current Thoughts (Centos 5.3 Platform) I'm looking into doing an HA setup for a Asterisk 1.4 install on Centos. I've seen a number of different pointers to packages for this some of which are packages that seem quite dated from an update perspective (Ultra Monkey links I've seen haven't been updated in a while). What is the current best practice on this for this platform? My first foray into any of the Linux HA setups but not afraid of the command line. Jim ___ -- 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] Asterisk on a Beagleboard?
- Original Message - From: Martin asteriskl...@callthem.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, September 23, 2009 11:01:04 AM GMT -05:00 US/Canada Eastern Subject: Re: [asterisk-users] Asterisk on a Beagleboard? Even PCI has 133MB/s ... so what ? Also isn't USB only target ? It doesn't do DMA ... so it might be same as PCI Target chips that slow down the CPU TDMoE has to have those frames on time all the time forever ... these ethernet frames are sent both ways every 1ms that might be (or not) too much load on the small CPU loose a few frames or deliver late and your voice TDMoE won't work right I just speculate here Martin On Wed, Sep 23, 2009 at 7:56 AM, Jeff LaCoursiere j...@jeff.net wrote: On Wed, 23 Sep 2009, Tzafrir Cohen wrote: On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote: I do not know if fonebridge would work here since it sends/receives the ~2 Mbps (for each circuit/port) of data over ethernet ... constantly. That could choke the USB ... Ethernet has frames. While I'm not exactly sure how ethernet over USB works and how TDM over Ethernet (MF) works, I would speculate that it is far from flooding the USB bus. Even USB 1.1 was 12Mbps. Should be plenty of room for a mere 24 channels of ulaw :) j The test we did was actually with 2x T1s worth of calls (48 uLaw calls) on the Beagleboard using the Dual port fonebridge. I'm not suggesting this would be a good production quality system. I think a native Ethernet connection and not via a USB adapter would be more efficient but the CPU was able to handle the call volume no problem. ___ -- 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] Asterisk on a Beagleboard?
Yes. Using Ubuntu, Asterisk with Dahdi. USB to Ethernet HUB and a Redfone fonebridge T1/E1 gateway connected to it. It can process a T1/E1 worth of calls no problem. - Original Message - From: Vincent vincent.delpo...@bigfoot.com To: asterisk-users@lists.digium.com Sent: Tuesday, September 22, 2009 8:56:55 AM GMT -05:00 US/Canada Eastern Subject: [asterisk-users] Asterisk on a Beagleboard? Hello Out of curiosity, has someone managed to run Asterisk on a Beagleboard for home-use? www.beagleboard.org As an alternative to a PC, it can be powered from a USB hub, so that would make for a compact, fanless Asterisk server. Thank you. ___ -- 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] External PRI Appliance
Try this for understanding the HA piece of the puzzle. http://support.red-fone.com/downloads/elastix/Elastix_HA_Cluster.pdf - Original Message - From: Frank Bulk frnk...@iname.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, June 10, 2009 4:22:31 PM GMT -05:00 US/Canada Eastern Subject: Re: [asterisk-users] External PRI Appliance It's not clear where the HA comes in. Can you explain? Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrin Henshaw Sent: Wednesday, June 10, 2009 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] External PRI Appliance Hello, I'm supporting an Asterisk setup in the Cayman Islands(I'm in Canada), which is running a PRI to a local telecom provider. We are looking at improving the setup, setting up high availability etc. My manager is interested in putting a TDMOE device in place, so we can easily switch the line remotely. He's looking at the following device: http://www.red-fone.com/index.php?page=shop.product_detailsflypage=flypage. tplproduct_id=26category_id=6option=com_virtuemartItemid=55 Anyone have any experience with this device? I'm interested in success/horror stories on it. Thanks. This email and its attachments may be confidential and are intended solely for the use of the individual or parties' to whom it is addressed. All comments are solely those of the author and do not necessarily represent those of Ignition. If you are not the intended recipient of this email and its attachments, you must take no action based upon them, nor must you copy or show them to anyone. Please contact the sender if you believe you have received this email in error. Thanks for considering the environmental impact before printing this email. ___ -- 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 ___ -- 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] COSTA RICA - E1
Yes. We have a number of customers in CR connecting to E1 PRIs using the Redfone fonebridge and it works fine. Are you having a particular issue or just looking for general confirmation that Asterisk and E1 in Costa Works? Good luck. - Original Message - From: Luis Morales faston...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 24, 2009 9:16:05 AM GMT -05:00 US/Canada Eastern Subject: [asterisk-users] COSTA RICA - E1 Does any have experience with E1 telephony support plus asterisk in costa rica ? Regards, Luis Morales -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- 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] E1/T1 failover hardware
We use the following device for Asterisk fail-over and our T1s. I believe they have an E1 version also: http://www.red-fone.com/fonebridge.html On Thu, 2005-10-20 at 11:23, John Daragon wrote: Warning ! I know zip about electronics. I've been looking for a device to handle the switching of an E1 connection from one Asterisk box to another in the event of a catastrophic server failure. All of the solutions I've seen so far have been designed to handle the situation where the telco line faults so that the local PBX can switch to a secondary E1. I've come across this application note : http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857 which describes T1/E1/J1, N+1 Redundancy With Analog Switches These parts are obviously designed to be built into E1 boards - hence, I think, the protection circuitry. Here's the question, then : what (apart from jumping through regulatory hoops) is to stop a simple array of MOSFETS (and a bit of control circuitry) implementing a failover switch controlled (say) by a pin on a serial or parallel port ? jd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] web management interface
Common requests from my customers include; -MACs (moves,adds,changes) on extensions (sip, zaptel,CID) -Voice Prompt recording/modifying -CDR Access on the fly -Reboot/halt option -The Multi-tenant functionality would be very nice also.Big market for that. Hope this helps. Good luck! On Wed, 2005-10-26 at 13:59, snacktime wrote: I'm finishing up a first version of a web interface for end users. It's focus is specific for our own uses, but I plan on releasing it under an open source license and would appreciate any feedback while I wrap up the first version. The interface is designed for end users without any real technical knowledge of asterisk except for some basic concepts of how things relate to each other. Such as contexts in a dialplan and how they relate to the context assigned to a sip/iax user, etc.. The interface is for day to day management of areas such as the dialplan and configuring new providers and phones in sip.conf and iax.conf. Things that an end user would want to change on their own. It also includes a nice voicemail interface for voicemail users, and some ability to manage/monitor asterisk via the manager api. One of the main features is the ability to write canned scripts that have associated configuration pages. A script is a text file with the script, and a YAML definition file. In the text file you can put variable placeholders, and in the YAML file you define the variables. The web interface then builds an html form based on the text file and the YAML definition. This way it's easy to add configurable sections in extensions.conf without having to change any of the base code. For instance providing canned scripts for extensions, call routing, voice menu's, etc.. If you have a script that needs a more custom web interface you can do that also by just creating the html form by hand. The same template approach is also used for configuring phones. Since we will be using this for local and remote installations, we also needed multi tenant capability. A basic multi tenant feature set is built in, so multiple businesses can be maintained on one copy of asterisk. Another requirement we had is to be able to coexist with an existing asterisk installation, instead of requring that the management interface take over all the asterisk config files. All you have to do with asterisk is add one include line in each .conf file you want to manage. And last but not least, another reason we couldn't use any of the existing interfaces is that almost without exception all of them were too difficult to install. Or more correctly unnecessarily difficult. We need to have something we can hand our clients and know they will be able to install the thing and run it with little difficulty. Since this interface uses ruby on rails, it includes a built in webserver, and the installation is a matter of untarring the distribution into a directory, changing the ownership of the directory to something asterisk can read, and running the start script to bring up the webserver. If we can work out a bug in tar2rubyscript that makes it fail on freebsd, then the distribution will be just one single executable that you can run as is. I would be very interested in hearing about what features people would like in a tool like this. Keeping in mind that it's not a complete asterisk system and is designed to work with existing installations. I will post a live demo in the next week or so once we get the first release ready. Chris __ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to configure the communication between two Asterisk servers
Tielin Xu wrote: Hi All: I have special set up to be done. See anyone can help me some ideas. Two Asterisk servers, server A trunks to PSTN, server B works as call routing engine. All sip phones are registered in server B. I have scenario like following: 1. A call comes to server A, server A sends the call related information to server B, assume that uses fast AGI. 2. Server B receives the message from server A, and look up dial plan for call routing, 3. Serve B sends the extension number back to server A, 4. Server A routes the call to the assigned agent. How does server B receive the message from server A? Many thanks for your help. Tielin Xu CTI Analyst Nintendo of America ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You can trunk your two servers through IAX or similar but I sense you are looking for something that goes beyond that though it's not too easy to discern from your messageWhy not have server B route the calls to the SIP agents registered on the same server B? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE and Badness in Kernel
I'm going to poll the group one more time on this one. I have posted this before and didn't get any takers. Digium advises that I should just do IAX in place of TDMoE but I don't have that luxury. I have a very complex dial plan built around the TDMoE functionality and it would be very difficult/expensive to rewrite it. This has always worked excellent on 2.4 but now that we need to upgrade to 2.6 I'm getting all kinds of headaches. I'm willing to pay a consultant to work this out for me. Please contact me off list if interested The following is my original message: Badness in local_bh_enable at kernel/softirq.c on 2.6.X I'm seeing this on Kernel 2.6.+ implementations, namely Centos 4.1, FC4 machines while trying to do TDMoE trunks between two machines. 2.4 Kernel operates fine on the same hardware I'm compiling zaptel-1.0.9.2 as per instructions in README.Linux26 + README.udev. I've also tried CVS head zaptel. Here are some references where the issue has been reported before but I've yet to find a documented solution; http://lists.digium.com/pipermail/asterisk-users/2005-February/091867.html http://bugs.digium.com/view.php?id=5126 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soekris and Asterisk
You should look at the Redfone fonebridge product. I believe their product does what you are wanting to do; http://www.red-fone.com/fonebridge.html On Tue, 2005-10-11 at 00:38, Craig Guy wrote: Has anyone on the list used a Soekris engineering PC as a TDM - Ethernet bridge? For example something like a net4801 with a TE110p in it and then using TDMoE to get it into a bigger server where the call processing proper will occur. Anyone know if it might handle a quadspan card ok? (no transcoding, just pure PRI to TDMoE bridging). Craig ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMOE Badness in kernel...
I'm seeing the same behavior on a Debian system with 2.6.12. I have two systems with Digium Quad T1s in each and I trunk them with TDMoEThis always worked great on 2.4 and up to 2.6.8 but beyond that it either spits out copious amounts of kernel badness and paralyzes the system completely or gives the same results as you mentioned with constant Alarms/Alarm Clears. I've mentioned it on this forum before as well. On Wed, 2005-10-05 at 09:49 -0700, [EMAIL PROTECTED] wrote: Has anyone succesfully used TDMoE on Fedora Core 2.6.12+ Kernel versions? I'm having the issue that is in the Mantis bug database with badness with the kernel. My Story: I can get the dynamic span to come up and show OK in the zttool on both machines. However i get errors every second (Warning: detected alarm on channel 1... then channel 2...) And then the next second, i get : alarm cleared on channel 1,... channel 2... etc... No call will go through across the link becuase of the alarms. It looks as if 2.4 Kernel works, but it would be a lot of work to go back in time. Can anyone give me some direction on this. I have setup IAX2 between the two machines, but I would like the ability to use Dial(Zap/group number/Exten) I havent found a solution in reading through the wiki's about doing something similar with IAX.. i.e (Dial/IAX2/group number/$exten)... There are some scripts and macros that require you to code variables and check status of each trunk etc but it would be nice to use a group with IAX, and in the IAX.conf place iax in groups... (unless i just havent' found it)... Help? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE + kernel badness
Badness in local_bh_enable at kernel/softirq.c on 2.6.X I'm seeing this on Kernel 2.6.+ implementations, namely Centos 4.1 machines while trying to do TDMoE trunks between two machines. 2.4 Kernel operates fine on the same hardware I'm compiling zaptel-1.0.9.2 as per instructions in README.Linux26 + README.udev Here are some references where the issue has been reported before but I've yet to find a documented solution; Any tips? http://lists.digium.com/pipermail/asterisk-users/2005-February/091867.html http://bugs.digium.com/view.php?id=5126 Thanks. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: equipment configuration help
Erick Perez wrote: So, with this i solve the issue on main office. But what about the two remote? they are so little that they will not let me place another * box there. The phones will be SIP and they are like this INTERNET--PIX--LAN(machines and sip phones). The pixes in those two offices have an ipsec tunnel with the main office via internet. I was thinking of placing the asterisk with a public IP so the remote phones can NAT outside to the public asterisk located in the main office. What do you think? On 9/2/05, asterisk groups [EMAIL PROTECTED] wrote: That is correct. Normally the layer 3 switches include advanced features such as QoS but they may be available on simpler layer 2 switches. I think the key words to look for are 'Managed, QoS (802.1p) with priority queues, VLAN, (802.1q)'...maybe even PoE if you go with some SIP phones in the future that can be powered by Power Over Ethernet. Something else to keep in mind. best of luck. On Thu, 2005-09-01 at 22:03 -0500, Erick Perez wrote: Why an L3? just for the QoS part? I checked the alliedtelesyn 8624T at $1000.00 http://www.cdw.com/shop/products/default.aspx?EDC=772793 but i also looked at the 8550T which has 48 port 10-100 but L2 http://www.cdw.com/shop/products/default.aspx?EDC=773964RecommendedForEDC=772793RecoType=upsell at 900.00 is the QoS different? sorry for the question but i keep reading that asterisk needs qos to function better. Thanks, On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick- Can't say if they will or not. In theory they should respect all outgoing traffic unless being filtered by another device such as your PIX. You might want to check with the ADSL router manufacturer just to be safe. On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote: Do i have to change the adsl routers? or just do QoS with the Layer 3 switches? Will my ADSL router respect the QoS setting when sending the packet to the Internet? On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, After reviewing your original message a little closer it occurs to me that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400 cards. These are Quad FXS or FXO cards that could receive the lines from your 8 analog line card. You'll still need an E1 card (Digium or Sangoma) to terminate your PRI line, but maybe with those TDM400 cards you can avoid the added cost of a channel bank. Regarding your WAN and branch offices; 1. I've seen comments that tunneling VoIP traffic through IPSec can add overhead/delay that could impact voice quality. Something to keep in mind. I have trunked Asterisk boxes in MIA,BUE,SAO, etc. but trunk with IAX over the Internet not tunneled or encrypted and performance is fine. 2. In your two locations with 15 50 users you should consider installing Asterisk boxes in those locations and trunking them together with IAX over the Internet. Perhaps go ahead and do the same thing with the smaller office. You can justify a small Asterisk implementation in an office with 5 phones. 3. For QoS look for L3 managed switches that can do QoS and/or bandwidth allocation. Cisco, Dell, Nortel, HP can all do this, maybe even more economical D-Links. Put these behind your PIX. It is also recommended to do separate VLANs for any SIP hard phones you deploy. This adds another layer of security and reliability. Hope this helps. On Wed, 2005-08-31 at 21:43 -0500, Erick Perez wrote: -M, The norstar has no E1 card, i will have to ask the nortel provider for the cost of it and configuration prices. I might end up paying the same as the channel bank. I was also thinking of using a Citel SIP-N-NORSTAR converter but its priced at around 3k. Too expensive because its only 24 ports and i have 32 nortel phones. According to this wiki http://www.voip-info.org/tiki-index.php?page=Asterisk+Nortel One problem with this approach is that in a Norstar system, it isn't easy to forward an extension to an outside line, which means Norstar phone users will have to remember to do something different when they want to call a user who has been switched to an IP phone for example. I guess that can be sorted out. Any manuals out there for configuration like [Telephone Company] ---E1--- [Asterisk with sangoma s102] ---T1 channel bank--- [Norstar]? (only the asterisk-t1-norstar part) Now another section, networking. The 3 offices are linked via VPNs like this Internet---ADSL Router-Cisco PIX Firewall---LAN doin ip tunneling will solve all communication problems internally, but what about QoS and SIP phones being taken to the public internet? one office has 5 users, the other 15, the other 50. ADSL Router recommendiations? and as for the phones being taken to the outside? what kind of configuration do i use? IAX is not an option. On 8/31/05, asterisk groups [EMAIL PROTECTED] wrote: Erick, Consider trunking your Meridian to the Asterisk via an E1 card on the Nortel.
