Re: [asterisk-users] fail-over server
On Wed, Feb 9, 2011 at 6:55 AM, Vieri rentor...@yahoo.com wrote: [snip] Since all of the SIP devices in my LAN have static IP addresses, I can keep track of everyone on my own. For instance, could I do fake SIP registrations from localhost (the * server) and specify a LAN IP address? Have you looked at the 'defaultip' sip configuration option? Or setting host=IP for those devices? I would write a custom script that would execute whenever an Asterisk server takes over. As said earlier, this server would not have any SIP extensions registered at first and they would be registering slowly within 60 seconds or more. However, since I KNOW FOR SURE that some SIP devices are always online and have static IP addresses, can't I fool Asterisk by somehow registering via locahost but spoofing the source IP address? Maybe setting the source port to what it was exactly can be tougher but I *could* try to keep track of it. That sounds more complicated and likely to break than using Realtime. This way, whenever the Asterisk server that took over tries to bridge a call, it will try to connect to the fakely-registered IP address. I'm not using realtime for 2 reasons: 1- I'm using the FreePBX framework and there's no realtime backend unfortunately. Moving to Realtime and losing all the FreePBX goodies is time-consuming. Does anyone know how to use FreePBX + Realtime? This is unfortunate for most of the Asterisk GUI's available. 2- I don't have enough hardware resources to setup a server for the realtime DB that both Asterisk servers would connect to. Also, I wouldn't feel comfortable having just one DB server. For easier maintenance I would use a clustered database for realtime. However, I'm using Mysql 5.0 ndbcluster tables for other non-voip purposes and my experience hasn't been so great. I once had a power outage and all ndb table data was lost. Also, 5.0 ndb crashes in several occasions. As far as I can tell, it isn't reliable. I haven't tried 5.1 though. I have no experience with clustered postgresql. So run the DB on the same server as Asterisk, if your call volume allows it, and either replicate the data using the built-in DB replication or use DRBD between the two existing servers. We use DRBD between two Asterisk nodes on smaller installations for configurations and voicemail. It works very well for us. For MySQL Cluster to work well, the application has to be designed for it, and it is a RAM based storage. But that is a conversation for another list. -Jonathan -- _ -- 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] fail-over server
--- On Thu, 2/10/11, Jonathan Thurman jonat...@thurmantech.com wrote: Have you looked at the 'defaultip' sip configuration option? Or setting host=IP for those devices? I've read that defaultip can only be used on type=peer and when host=dynamic. I use type=friend. host=IP seems to be OK for me. I actually tried this option some time ago but had trouble with something I can't recall right now so reverted to dynamic. I guess I'll have to give it another shot. I'll try that before migrating to realtime... Thanks Jonathan! -- _ -- 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] fail-over server
--- On Tue, 2/8/11, Jonathan Thurman jonat...@thurmantech.com wrote: It depends on your configuration. If you use Asterisk Realtime to store SIP registrations, then the database will contain information on how to contact the device (fullcontact, ipaddr, and port fields). Then on a failover, Asterisk will do a lookup for the peer in the database, find the needed information and dial the device. I don't use realtime and haven't tried it yet. I don't know much about the SIP protocol but can't the server send a notification of some sort to peers so as to quicken re-registration? I'm thinking of something similar to sip notify. Since all of the SIP devices in my LAN have static IP addresses, I can keep track of everyone on my own. For instance, could I do fake SIP registrations from localhost (the * server) and specify a LAN IP address? I would write a custom script that would execute whenever an Asterisk server takes over. As said earlier, this server would not have any SIP extensions registered at first and they would be registering slowly within 60 seconds or more. However, since I KNOW FOR SURE that some SIP devices are always online and have static IP addresses, can't I fool Asterisk by somehow registering via locahost but spoofing the source IP address? Maybe setting the source port to what it was exactly can be tougher but I *could* try to keep track of it. This way, whenever the Asterisk server that took over tries to bridge a call, it will try to connect to the fakely-registered IP address. I'm not using realtime for 2 reasons: 1- I'm using the FreePBX framework and there's no realtime backend unfortunately. Moving to Realtime and losing all the FreePBX goodies is time-consuming. Does anyone know how to use FreePBX + Realtime? 2- I don't have enough hardware resources to setup a server for the realtime DB that both Asterisk servers would connect to. Also, I wouldn't feel comfortable having just one DB server. For easier maintenance I would use a clustered database for realtime. However, I'm using Mysql 5.0 ndbcluster tables for other non-voip purposes and my experience hasn't been so great. I once had a power outage and all ndb table data was lost. Also, 5.0 ndb crashes in several occasions. As far as I can tell, it isn't reliable. I haven't tried 5.1 though. I have no experience with clustered postgresql. In the above scenario, I can kill Asterisk, start it again, and place a call from two devices that have not registered again. I'd like to do that without Realtime (or with Realtime+FreePBX) or with any other means that doesn't require more than 2 servers (2 asterisk boxes)? Feedback appreciated. Thanks Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 -- _ -- 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] fail-over server
On 2/9/11 6:55 AM, Vieri wrote: I'd like to do that without Realtime (or with Realtime+FreePBX) or with any other means that doesn't require more than 2 servers (2 asterisk boxes)? we use drbd nfs cluster to store asterisk's ASTDB voice mail files but that would involve installing 2 extra servers for such purpose. however you can look into csync2 to sync all asterisk files between the 2 asterisk servers if you don't want extra hardware. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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] fail-over server
Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually on-line so calls can be routed to them. How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri -- _ -- 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] fail-over server
Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote: How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? If you have two identical servers online, it is better to make a HA sollution. Sorry, I haven't made HA Asterisk yet, I can not help more. -- Best regards, Gergomailto:csi...@gmail.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] fail-over server
Take a look at High Availability ASTerisk (HAAST) from www.generationd.com Their software sits between the OS and asterisk, and can failover servers, switch IP addresses, control external interfaces, etc. It can run on different hardware (make a cluster from different/cheap boxes), it allows long distance seperation of cluster members, etc. Also, it's easy to install. Michelle (I'm affiliated with generationd so I may be biased, but I think the product is awesome) From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Gergo Csibra [csi...@gmail.com] Sent: Tuesday, February 08, 2011 11:17 AM To: Asterisk Users List Subject: Re: [asterisk-users] fail-over server Tuesday, February 8, 2011, 5:07:29 PM, Vieri wrote: How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? If you have two identical servers online, it is better to make a HA sollution. Sorry, I haven't made HA Asterisk yet, I can not help more. -- Best regards, Gergomailto:csi...@gmail.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 -- _ -- 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] fail-over server
Hi, Thats very simple. Use sip realtime registration with mysql and heartbit to control switiching. Regards, Carlos M Cruz Em 2011/02/08 16:07, Vieri rentor...@yahoo.com escreveu: Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually on-line so calls can be routed to them. How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? Thanks, Vieri -- _ -- 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
Re: [asterisk-users] fail-over server
On Tue, Feb 8, 2011 at 8:07 AM, Vieri rentor...@yahoo.com wrote: Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. This is a typical setup for two node HA. Just be careful when clustering only two servers. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually on-line so calls can be routed to them. It depends on your configuration. If you use Asterisk Realtime to store SIP registrations, then the database will contain information on how to contact the device (fullcontact, ipaddr, and port fields). Then on a failover, Asterisk will do a lookup for the peer in the database, find the needed information and dial the device. Of course any registrations that happen before being written right before the server fails may not work. Also make sure to use the latest version of Asterisk as there was a bug where fullcontact wasn't saved correctly. How can I minimize this time lapse? Can Asterisk notify all SIP clients in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)? In the above scenario, I can kill Asterisk, start it again, and place a call from two devices that have not registered again. So, the best timeout is your dead time detection and failover startup time. -Jonathan -- _ -- 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] Fail over to Pri on VoIP connection failure
All, Thanks for the help. Checking on and changing the route based on dialstatus is the way to go. Thanks, ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
; call $ARG1 through nufone, failing over to the PRI.
[macro-nufone-dial]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],,go)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Congestion
exten = dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
It really is as simple as that. :-)
-A.
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RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure
Andrew,
Thanks for this - I have also been looking for a way to fail over
calls to a second SIP path, but;
In the event that the first attempt DOES NOT RESPOND (is down) there has
to be a timeout value to go to the next priority, correct? Otherwise the
channels just sits silent waiting for a response.
I think your macro assumes that you got a response from nufone, but what
if they were dead in the water?
Have I missed something?
Is there a way to modify the relevant SIP timer so if the INVITE is not
ack'd in a specific period of time then the next priority is executed?
Damon
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
Sent: Friday, January 27, 2006 1:12 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection
failure
On Thursday 26 January 2006 10:52, Cavanna, Richard wrote:
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas
calls(busy
or just a hang-up). Is there a way in the dial plan to
automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
; call $ARG1 through nufone, failing over to the PRI.
[macro-nufone-dial]
exten = s,1,Dial(SIP/[EMAIL PROTECTED],,go)
exten = s,n,NoOp(NUFONE: HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS
is
${DIALSTATUS})
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Congestion
exten = dial-CHANUNAVAIL,1,Macro(pri-dial,${ARG1},${ARG2})
It really is as simple as that. :-)
-A.
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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
On Friday 27 January 2006 16:00, Damon Estep wrote: In the event that the first attempt DOES NOT RESPOND (is down) there has to be a timeout value to go to the next priority, correct? Otherwise the channels just sits silent waiting for a response. That's what the qualify parameter in sip/iax.conf is for. Never terminate calls without it. :-) It won't *guarantee* that you'll never get dead air, but it sure goes a long way to ensuring that it happens so infrequently you'll think you misdialed. I think your macro assumes that you got a response from nufone, but what if they were dead in the water? Then qualify would have failed and Dial() would have immediately returned CHANUNAVAIL. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, January 27, 2006 2:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure On Friday 27 January 2006 16:00, Damon Estep wrote: In the event that the first attempt DOES NOT RESPOND (is down) there has to be a timeout value to go to the next priority, correct? Otherwise the channels just sits silent waiting for a response. That's what the qualify parameter in sip/iax.conf is for. Never terminate calls without it. :-) It won't *guarantee* that you'll never get dead air, but it sure goes a long way to ensuring that it happens so infrequently you'll think you misdialed. I think your macro assumes that you got a response from nufone, but what if they were dead in the water? Then qualify would have failed and Dial() would have immediately returned CHANUNAVAIL. -A. OK - starting to make sense now Qualify=yes for the peer in sip.conf If you have qualify=yes I assume that triggers a sip query to get channel capabilities from the peer? What is the qualify timeout? Can it be manipulated? If the goal was strictly to try one provider, and if the channel fails qualify, then try the next, is the macro you posted needed? Couldn't you just; Exten = ,1,Dial(SIP/[EMAIL PROTECTED] Exten = ,2,Dial(SIP/[EMAIL PROTECTED] Exten = ,3,Congestion(15) Exnte = ,4,Hangup ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
On Friday 27 January 2006 16:24, Damon Estep wrote: If you have qualify=yes I assume that triggers a sip query to get channel capabilities from the peer? What is the qualify timeout? Can it be manipulated? qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for a response. If it does not receive one within 1000ms (by default) and qualifysmoothing is not enabled, it will flag the peer as UNREACHABLE which means that any attempts to Dial() the peer will fail immediately with CHANUNAVAIL. Asterisk continues to send these pings until it receives a response within the accepted timeframe and once it gets responses again it will flag the peer as being available once again. There are some other tuning parameters which can be used to modify this behaviour slightly but this is what qualify does in a nutshell. If the goal was strictly to try one provider, and if the channel fails qualify, then try the next, is the macro you posted needed? Correct. Couldn't you just; Exten = ,1,Dial(SIP/[EMAIL PROTECTED] Exten = ,2,Dial(SIP/[EMAIL PROTECTED] Exten = ,3,Congestion(15) Exnte = ,4,Hangup Well I've never been a fan of just letting things fall off the edge and expecting them to work reliably. I use the 'g' Dial() option so that I can handle failover and call completion correctly or properly -- instead of just letting it do whatever svn trunk deems right at this point I specifically do things based on how the call terminated. It's just a nicer way of doing what you've provided, and ends up being more robust to code policy changes. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail over to Pri on VoIP connection failure
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Friday, January 27, 2006 2:45 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure On Friday 27 January 2006 16:24, Damon Estep wrote: If you have qualify=yes I assume that triggers a sip query to get channel capabilities from the peer? What is the qualify timeout? Can it be manipulated? qualify (for SIP) sends a SIP OPTIONS packet to the peer and waits for a response. If it does not receive one within 1000ms (by default) and qualifysmoothing is not enabled, it will flag the peer as UNREACHABLE which means that any attempts to Dial() the peer will fail immediately with CHANUNAVAIL. Asterisk continues to send these pings until it receives a response within the accepted timeframe and once it gets responses again it will flag the peer as being available once again. There are some other tuning parameters which can be used to modify this behaviour slightly but this is what qualify does in a nutshell. Since your original hint on qualify=yes have been hunting for the parameter tuning capabilities of this feature - to no avail. Are you aware of any reference anywhere on tuning the qualify frequency and timeout? I assume this (tuning) does not require code changes. Correct? If the goal was strictly to try one provider, and if the channel fails qualify, then try the next, is the macro you posted needed? Correct. Couldn't you just; Exten = ,1,Dial(SIP/[EMAIL PROTECTED] Exten = ,2,Dial(SIP/[EMAIL PROTECTED] Exten = ,3,Congestion(15) Exnte = ,4,Hangup Well I've never been a fan of just letting things fall off the edge and expecting them to work reliably. I use the 'g' Dial() option so that I can handle failover and call completion correctly or properly -- instead of just letting it do whatever svn trunk deems right at this point I specifically do things based on how the call terminated. It's just a nicer way of doing what you've provided, and ends up being more robust to code policy changes. Sounds like words of wisdom to me :) Thanks a million D ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fail over to Pri on VoIP connection failure
I am trying to tweak my dial plan and I am running into a problem.
Sometimes my VoIP out bound calls do not complete on overseas calls(busy
or just a hang-up). Is there a way in the dial plan to automatically
dial out of my PRI when something like this happens. Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the innocent)
-- Executing Dial(Zap/47-1,
IAX2/VoIPServicePrividerOUT/011) in new stack
-- Called VoIPServicePrividerOUT/011
-- Call accepted by 72.34.43.5 (format g729)
-- Format for call is g729
-- Channel 0/23, span 2 got hangup request here I get a busy
signal
-- Hungup 'IAX2/ VoIPServicePrividerOUT-1'
[Outbound context]
exten = _9011.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9011.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9011.,3,Macro(outisbusy) ; No available circuits
exten = _918.,1,Macro(dialout-trunk,2,${EXTEN:1},); 800 numbers to the
PRI
exten = _918.,2,Macro(outisbusy) ; No available circuits
exten = _9Z.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9Z.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9Z.,3,Macro(outisbusy); No available circuits
Richard
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Re: [Asterisk-Users] Fail over to Pri on VoIP connection failure
I know this may be a backwards way but for several
reasons I have asterisk send all calls thru astcc.
With astcc you specify multiple routes with prioroty
settings. If it cant complete a call with one route it
will roll over and use the next one.
Regards,
Dovid
--- Cavanna, Richard [EMAIL PROTECTED] wrote:
I am trying to tweak my dial plan and I am running
into a problem.
Sometimes my VoIP out bound calls do not complete on
overseas calls(busy
or just a hang-up). Is there a way in the dial plan
to automatically
dial out of my PRI when something like this happens.
Either by time
limit by a failure event?
Any point in the right direction would be great
Thanks,
CLI output (cleansed to protect the innocent)
-- Executing Dial(Zap/47-1,
IAX2/VoIPServicePrividerOUT/011) in
new stack
-- Called VoIPServicePrividerOUT/011
-- Call accepted by 72.34.43.5 (format g729)
-- Format for call is g729
-- Channel 0/23, span 2 got hangup request
here I get a busy
signal
-- Hungup 'IAX2/ VoIPServicePrividerOUT-1'
[Outbound context]
exten = _9011.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9011.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9011.,3,Macro(outisbusy); No available
circuits
exten = _918.,1,Macro(dialout-trunk,2,${EXTEN:1},);
800 numbers to the
PRI
exten = _918.,2,Macro(outisbusy) ; No available
circuits
exten = _9Z.,1,Macro(dialout-trunk,4,${EXTEN:1},)
exten = _9Z.,2,Macro(dialout-trunk,2,${EXTEN:1},)
exten = _9Z.,3,Macro(outisbusy) ; No available
circuits
Richard
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Re: [Asterisk-Users] Fail over using CHANAVAIL
On Sunday 22 January 2006 14:11, Chris Mason wrote:
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but this returns the provider but not the username, so I
don't understand how to use this for real applications to determine IAX2
availability. The only way I can see to use it is to only specify one
channel and test it, jumping to n+101 if it isn't.
That is pretty much how I do things. I use qualify for my SIP and IAX2
connections and then basically do something like this:
In my nufone-dial macro():
exten = s,n,Dial(IAX2/[EMAIL PROTECTED]/${ARG1},,go)
exten = s,n,Goto(dial-${DIALSTATUS},1)
exten = dial-CANCEL,1,Hangup
exten = dial-ANSWER,1,Hangup
exten = dial-NOANSWER,1,Hangup
exten = dial-BUSY,1,Busy
exten = dial-CONGESTION,1,Congestion
exten = dial-CHANUNAVAIL,1,Macro(asterlink-dial,${ARG1},${ARG2})
And then the asterlink-dial macro is almost identical, except CHANUNAVAIL
calls the pri-dial macro, which I use as a last-effort attempt to get a call
out, as it's my most expensive route.
-A.
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RE: [Asterisk-Users] Fail over using CHANAVAIL
I am trying to construct a macro for long distance dialling. I have
two internet feeds, I have all routes including Teliax on
Internet A
and a static route to Voxee on Internet B.
Here's an AEL macro I use on our boxes. Modify for your needs.
// dial a number with a range of routing options
macro outbound (number, clid, route1, route2, route3, route4) {
if (${clid} = ) {
CALLERID(number)=${DEFAULTCID};
} else
CALLERID(number)=${clid};
dialstart:
switch (${route1}) {
case direct:
dialout (${number});
break;
case providera:
dialout (IAX2/providera/${number});
break;
case providerb:
dialout (IAX2/providerb/${number});
break;
case providerc:
dialout (SIP/[EMAIL PROTECTED]);
break;
case pstn:
dialout (SIP/[EMAIL PROTECTED]);
break;
default:
NoOp (invalid route: ${route1});
};
route1=${route2};
route2=${route3};
route3=${route4};
if (${route1} = ) {
Playtones (info);
Congestion ();
};
goto dialstart;
};
// dial a number ignoring anything except busy
macro dialout (dialstring) {
Dial (${dialstring},,TW);
switch (${DIALSTATUS}) {
case BUSY:
Playtones (busy);
Busy ();
break;
};
};
Basically, replace dial commands in extensions.conf with a call to macro
outbound, passing it the number to dial, callerid to present, and any
number of routes in the order you want them to be tried. The macro dialout
just ensures that if the number called is genuinely busy, outbound doesn't
plough on with routes 2,3,4 regardless.
Hope that helps.
Regards,
Chris
--
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[Asterisk-Users] Fail over using CHANAVAIL
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the ChanIsAvail() command to check the
connection, but this returns the provider but not the username, so I
don't understand how to use this for real applications to determine IAX2
availability. The only way I can see to use it is to only specify one
channel and test it, jumping to n+101 if it isn't.
[globals]
[EMAIL PROTECTED]
[EMAIL PROTECTED]
[macro-longdistance]
;
; Standard extension macro:
; ${ARG1} - Number to dial
;
exten = s,1,SetCallerID(NetConcept1234567890|a)
exten = s,2,ChanIsAvail(IAX2/${TELIAX}IAX2/${VOXEE})
exten = s,2,Read(${AVAILCHAN})
exten = s,3,Cut(C=AVAILCHAN,,1)
exten = s,4,NoOp(AVAILCHAN= ${C})
exten = s,5,Dial(${C}/${ARG1},60,tr)
--
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Re: [Asterisk-Users] Fail over?
in extensions.conf exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,2,Dial(SIP/[EMAIL PROTECTED]) On 11/11/05, John E. Elkin [EMAIL PROTECTED] wrote: Maybe its already been posted, but i cant find it... I have an asterisk box running agilevoice (Customer signup and provisioning system) I have two sip termination providers. One provides did and termination. The other provides just my termination. My big question is. If the termination on provider A goes out.. i want my asterisk box to route calls to provider B how do i make this happen automaticly? John___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Fail over?
On Mon, 2005-11-14 at 13:11 -0800, Andy Kuo wrote: in extensions.conf exten = _X.,1,Dial(SIP/[EMAIL PROTECTED]) exten = _X.,2,Dial(SIP/[EMAIL PROTECTED]) I dont think that will work quite right starting with BRIStuff. While congestion() is +1 I believe if the peer is down its +201. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --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] Fail over?
Maybe its already been posted, but i cant find it... I have an asterisk box running agilevoice (Customer signup and provisioning system) I have two sip termination providers. One provides did and termination. The other provides just my termination. My big question is. If the termination on provider "A" goes out.. i want my asterisk box to route calls to provider "B" how do i make this happen automaticly? John___ --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] Fail over solutions
The disk array would be the only expensive add on, more than a normal asterisk system. It all depends on how important voicemail is in your application, although there are cheaper alternatives (NFS for example, but then your NFS server becomes a single point of failure, depending on the disk array that same issue could be true there as well). If you are on a budget, I would suggest to look at a drbd+heartbeat combination. DRBD is a block device which is designed to build high availability clusters. This is done by mirroring a whole block device via (a dedicated) network. You could see it as a network raid-1. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] Fail over solutions
On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ 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] Fail over solutions
One thing that could be done is to have a disk array for voicemail and all with dual controllers. Then plug that into each of two servers. Bind the IP components to a IP that is transportable between machines. When one fails ifconfig the failover machine to use that IP (could be a virtual interface). Veritas HA works similiarly that way. Via a serial cable there are 'global atomic broadcasts' basically a ping. If the ping fails to occur the machine marked backup assumes the IP for all services of the primary. Because it has access to the same disks it can mount them and carry on like nothing happened. Veritas seperates services from the machine. If you have say a web server, mail, and SIP you would have each one on a seperate IP so that if any one single service fails that one and only that one can be moved to the backup server. With asterisk this may be overkill. MAC addresses are the only other problem. Veritas accomplishes this by MAC spoofing. Cisco PIX do as well. You might, depending on specific ethernet driver, be able to ifconfig eth0 headdr 00:00:de:ca:fb:ad. Just a thought. On Wed, 2005-04-27 at 08:33 +0100, Jason Williams wrote: On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ 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 -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Fail over solutions
On Wed, 2005-04-27 at 00:52 -0700, trixter http://www.0xdecafbad.com wrote: One thing that could be done is to have a disk array for voicemail and all with dual controllers. Then plug that into each of two servers. Bind the IP components to a IP that is transportable between machines. When one fails ifconfig the failover machine to use that IP (could be a virtual interface). Veritas HA works similiarly that way. Via a serial cable there are 'global atomic broadcasts' basically a ping. If the ping fails to occur the machine marked backup assumes the IP for all services of the primary. Because it has access to the same disks it can mount them and carry on like nothing happened. Veritas seperates services from the machine. If you have say a web server, mail, and SIP you would have each one on a seperate IP so that if any one single service fails that one and only that one can be moved to the backup server. With asterisk this may be overkill. MAC addresses are the only other problem. Veritas accomplishes this by MAC spoofing. Cisco PIX do as well. You might, depending on specific ethernet driver, be able to ifconfig eth0 headdr 00:00:de:ca:fb:ad. Just a thought. I forgot to add that if you have T1/E1/J1s you would want a hunt group defined so that calls from one goto the other if the card is nonresponsive. Analogue lines can forward to a seperate machine on a 'no answer' basis. Of course if you are doing failover odds are you arent doing analogue lines. All in all this shouldnt be a terribly difficult solution to implement, and could even be done on 1U boxes or whatever. Basically a 'brain dead' add on package that requires little configuration, and then distributed by whatever means someone chooses (if they choose unwisely someone else will just write something similar that is distributed differently :) Due to the cost of asterisk this could be a feature that normal PBX systems do not have, or do not have for anything 'reasonably' priced. Giving yet another advantage to asterisk. The disk array would be the only expensive add on, more than a normal asterisk system. It all depends on how important voicemail is in your application, although there are cheaper alternatives (NFS for example, but then your NFS server becomes a single point of failure, depending on the disk array that same issue could be true there as well). -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Fail over solutions
Could you explain me some more how i could use dual controllers ? Is this done with special harddisks ? What hardware do i need to do this ? /Z. trixter http://www.0xdecafbad.com wrote: One thing that could be done is to have a disk array for voicemail and all with dual controllers. Then plug that into each of two servers. Bind the IP components to a IP that is transportable between machines. When one fails ifconfig the failover machine to use that IP (could be a virtual interface). Veritas HA works similiarly that way. Via a serial cable there are 'global atomic broadcasts' basically a ping. If the ping fails to occur the machine marked backup assumes the IP for all services of the primary. Because it has access to the same disks it can mount them and carry on like nothing happened. Veritas seperates services from the machine. If you have say a web server, mail, and SIP you would have each one on a seperate IP so that if any one single service fails that one and only that one can be moved to the backup server. With asterisk this may be overkill. MAC addresses are the only other problem. Veritas accomplishes this by MAC spoofing. Cisco PIX do as well. You might, depending on specific ethernet driver, be able to ifconfig eth0 headdr 00:00:de:ca:fb:ad. Just a thought. On Wed, 2005-04-27 at 08:33 +0100, Jason Williams wrote: On 4/26/05, snacktime [EMAIL PROTECTED] wrote: On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? The biggest problem with that solution is voicemail it could get left on one server and not be on the other for one hour. Jason ___ 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 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 signature.asc Description: OpenPGP digital signature ___ 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] Fail over solutions
On Wed, 2005-04-27 at 11:17 +0300, Zoa wrote: Could you explain me some more how i could use dual controllers ? Is this done with special harddisks ? What hardware do i need to do this ? We used a winchester drive array, which is not cheap, and way overkill for asterisk. EMC makes similar boxes. The one we had was a 19 inch cabinet and all drives were RAID. It came with integrated controllers each was dual ported so the machine could do 2x SCSI speeds, and there were 2 controllers integrated into the rack so both systems could benefit from this (ie 4 ports). I am unsure if there are smaller cheaper solutions, a multi-terabyte raid array would be underused for just voicemail unless you get a TON of voicemail, and I cant imagine asterisk being able to handle the clients that would require that. I would suggest googling multiport drive array I have not seen any ability to connect multiple controllers to the same disk, so you have to get a special controller that allows for this type of connectivity. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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] Fail over solutions
Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? Sean ___ 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] Fail over solutions
On 4/26/05, Sean Kennedy [EMAIL PROTECTED] wrote: Hi folks, I'm curious; What does everyone do for failover? I have two servers, same os/compilation. I designate one the master, the other the slave, and I rsync the config files once an hour and trigger a restart when convenient command on the console. These two servers are setup in the dns in a round robin fashion. What is everyone else doing? That's kind of a loaded question... Do you plan on expanding? What is your budget? What are your uptime requirements? Are you serving customers or is this just for internal use? Round robin dns is a cheap way of doing load balancing, not failover. If a server fails, you will still have requests going to the dead server. I always prefer to keep volatile data in a real database, and make sure that if anything is redundant, it's the database. You can survive downtime, but you might not survive the loss of critical data. Automate your backups, do them often, and keep a copy both on site and off site. Keep backups of data beyond what you think you will need them for at the time, because invariably you will need something that you deleted 6 months ago. If you are on a budget and want to just use what you have, then I would keep your same setup but dump the round robin dns. Chris ___ 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] Fail over
On 23:34, Tue 29 Mar 05, Mitchel Constantin wrote: Matt, This isn't meant as a flame, rather I'm curious about what other people think about the following situation...maybe it's just the philosopher in me, what happens when the load balancer fails? Good point. Was thinking the same thing. Why load balance with one machine ? This is where CARP would be great. But besides that, what happens when connectivity to this specific location goes down ? Only way to provide real HA is to use 2 seperate locations, like 2 different countries :) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ 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] Fail over
On Wed, 30 Mar 2005 05:03:33 +0800, El Flynn [EMAIL PROTECTED] wrote: Rich Adamson wrote: No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. Dataprobe makes a range of A/B switches, some with more intelligence that you might be able to use in this scenario. One of their products (check out http://www.dataprobe.com/switch/ab_net.html) has a feature which pings a specific IP address, and switches over once it stops getting a response. Some of their products are programmable too, where you can send TCP messages to initiate the switching process. Check out their website for more products. Flynn p/s I am in no way related to Dataprobe. This is just some stuff I received from them when asking a similar question on the list about six months ago. ___ 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 could use WhatsUp Gold or write your own code to monitor the Asterisk box. In WhatsUp, you can ping and/or define a custom device monitor. You custom device monitor can initiate a SIP or IAX conversation on the proper port if the response is correct all is well if not: Page me email me run an .exe 1) Running an .exe file that sends commands to an X10 appliance module(s). Turn off failed server, turn on standby. This requires boot time. 2) Running an .exe file that sends commands to the serial port. Attached to the serial port is a BASIC STAMP http://www.parallax.com/html_pages/products/basicstamps/basic_stamps.asp Buy the developers kit first time, then just the modules. It has 8 I/O ports that will control a 5v relay from RadioShack. No real skill needed here sample programs come with the kit. Program, attach serial cable. Second Stamp is dirt cheap. Jameco has them too. You could also attach the Stamp directly to the Asterisk box and run a cron job to tickle the BasicStamp to tell it I'm Alive, if the event fails, trigger your relays and switch your ethernet cables. The Stamps are really stable. Also the STAMPII and above will send X10 commands. Yet another choice is to get a PCI watchdog timer board. http://www.cyberresearch.com/store/pc-accessories-computer-parts/computer-accessories-pc-peripherals/watchdog-timers-wdt/WDT_PCIX_3118.2.htm http://www.berkprod.com/prices.htm Use them with relay to control ethernet or rely on them to reboot. -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ 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] Fail over
buy 2 load balancer to failover between themselves. Best Regards Matt - Original Message - From: Mitchel Constantin [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 11:34 PM Subject: Re: [Asterisk-Users] Fail over Matt, This isn't meant as a flame, rather I'm curious about what other people think about the following situation...maybe it's just the philosopher in me, what happens when the load balancer fails? Thanks, Mitchel On Tue, 29 Mar 2005 13:47:58 -0800, Matt [EMAIL PROTECTED] wrote: you can use dual T1, each on a separate pbx. and use a load balancer for fail over. see http://www.xgforce.com/loadbalancer.html for affordable models. Best Regards Matt - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 7:11 AM Subject: RE: [Asterisk-Users] Fail over No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. ___ 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 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 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 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] Fail over
For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ 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] Fail over
There's many solutions.. One being www.voiceguard.com I think might be what you want. - Original Message - From: Chris Mason [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 8:01 AM Subject: [Asterisk-Users] Fail over For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ 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 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] Fail over
No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Tuesday, March 29, 2005 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fail over There's many solutions.. One being www.voiceguard.com I think might be what you want. - Original Message - From: Chris Mason [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 8:01 AM Subject: [Asterisk-Users] Fail over For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ 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 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 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] Fail over
On Tue, 29 Mar 2005 09:40:08 -0400, Chris Mason [EMAIL PROTECTED] wrote: No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. The Sipura 3000 does this. That is what I use at home. -Brian ___ 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] Fail over
No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. ___ 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] Fail over
Rich Adamson wrote: No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. Dataprobe makes a range of A/B switches, some with more intelligence that you might be able to use in this scenario. One of their products (check out http://www.dataprobe.com/switch/ab_net.html) has a feature which pings a specific IP address, and switches over once it stops getting a response. Some of their products are programmable too, where you can send TCP messages to initiate the switching process. Check out their website for more products. Flynn p/s I am in no way related to Dataprobe. This is just some stuff I received from them when asking a similar question on the list about six months ago. ___ 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] Fail over
Some of their products are programmable too, where you can send TCP messages to initiate the switching process. Check out their website for more products. That's perfect, because I use a Nagios monitoring system that can tell if the Asterisk system is running and tell the fail-over switch to switch if it isn't. I'm not sure how to monitor Asterisk yet but it looks like this will do it: http://megaglobal.net/docs/asterisk/monitor_pbx.pl Together with that I would monitor disk space, cpu load, http and ping, should make sure everything is working well. The only other issue is power failure but with a large UPS system I don't expect that to be an issue. Chris ___ 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] Fail over
you can use dual T1, each on a separate pbx. and use a load balancer for fail over. see http://www.xgforce.com/loadbalancer.html for affordable models. Best Regards Matt - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 7:11 AM Subject: RE: [Asterisk-Users] Fail over No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. ___ 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 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] Fail over
Matt, This isn't meant as a flame, rather I'm curious about what other people think about the following situation...maybe it's just the philosopher in me, what happens when the load balancer fails? Thanks, Mitchel On Tue, 29 Mar 2005 13:47:58 -0800, Matt [EMAIL PROTECTED] wrote: you can use dual T1, each on a separate pbx. and use a load balancer for fail over. see http://www.xgforce.com/loadbalancer.html for affordable models. Best Regards Matt - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 7:11 AM Subject: RE: [Asterisk-Users] Fail over No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. ___ 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 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 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
