Re: [Asterisk-Users] Asterisk Dial Failover
On Fri, 2005-12-09 at 06:51 -0600, [EMAIL PROTECTED] wrote: > Your other option is to setup the OpenSER boxes in a truly redundant > configuration using Linux HA (www.linux-ha.org). That way you setup all > your PSTN calls to forward to one shared virtual IP between the boxes. One > of the boxes is the Master, the other is the Slave. There is a heartbeat > between the boxes that goes at a configurable rate. If the Master fails > then the Slave will take over and it can even be configured for sub-second > failover. I think there is a article on voip-info.org about this, but > don't have time to look it up. Not attempting to hijack the thread but does OpenSER/Linux-HA support stateful failover? If not wouldn't you be better off without the virtual IP address and phones that support a secondary proxy so the phone switches over the moment it detects that the primary is down? Regards, Patrick ___ --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] Asterisk Dial Failover
On 10:09, Fri 09 Dec 05, John Cianfarani wrote: > Ryan/Jonathan, > > Couple quick questions regarding your setup? > Do you operate this in a strictly master/slave setup? > Do you have anything(mon/ha's internal status/monitor options) that > actually monitors the asterisk process (to determine if it is hung). Or > is it only with total box failure to you fail over? > Do you use something to sync config/vm/cdr? Rsync/unison? Hi, We have the config/vm/cdr on a drbd device (part of linux-ha). We wrote a little script that checks the ps tree if asterisk is still there. It will bring down the interface asterisk is bind on, and the linux-ha heartbeat that monitors the ip's will set the slave to master and take over :) Greetz, -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ --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] Asterisk Dial Failover
WEll I personally have not implemented a Linux-HA cluster mainly because I don't have the resources to do so. I study Asterisk purley as a hobby (nerd.. yeahI know) because it is an awesome OSS product. Anyways, after some searching around I think it would not be TOO difficult to implement a resource failover using a combination of Linux HA (linux-ha.org), Mon (www.kernel.org/software/mon) and SIPp (sipp.sourceforge.net). Linux HA supports failover on resources or machine based failures(inherently if a machine goes down all the resources will be gone too). However Linux HA just provides the interface to say which machine is the "Active" machine in the cluster for a specific resource (aka Asterisk). That is where Mon would come in. Mon montiors services on a machine and can be configured to react if a service fails the periodic test. But Mon does not have a per-defined monitor script for Asterisk. That is where SIPp would come in. You could create a Mon script that calls SIPp and looks for the return code after a number of calls through the server. SIPp will return a 0 if all the calls succeeded. That way in your Mon script if SIPp returned anything other than a 0 then register it as a failure. Once a failure occurs you can configure Mon to switch the active Asterisk server using the Linux HA functionality. Like I said, there is turn key way of doing this, but it looks like a good little project for the wiki? Maybe I'll start working on this in my spare time, I just need to get some time to play with the different components. There are a few more logicistical things that would have to be taken care, mainly anything file realted, but that could be alleviated with some kind of remote mounted filesystem. Hope this helps, Ryan > Yes, that's a great question. I'm wondering the same thing. Can these > heartbeat apps monitor applications as well as network connectivity? The > heartbeat utility at www.linux-ha.org talks about monitoring some standard > apps like web servers and such but isn't clear about other apps... like > Asterisk or SER. > > -Original Message- > From: John Cianfarani [mailto:[EMAIL PROTECTED] > Sent: Friday, December 09, 2005 8:10 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Asterisk Dial Failover > > > Ryan/Jonathan, > > Couple quick questions regarding your setup? > Do you operate this in a strictly master/slave setup? > Do you have anything(mon/ha's internal status/monitor options) that > actually monitors the asterisk process (to determine if it is hung). Or > is it only with total box failure to you fail over? > Do you use something to sync config/vm/cdr? Rsync/unison? > > Thanks > John > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan > k. Creasy > Sent: Friday, December 09, 2005 8:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Asterisk Dial Failover > > I chose this method and have been happy with the results. > > -Jonathan > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > [EMAIL PROTECTED] > Sent: Friday, December 09, 2005 7:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk Dial Failover > > Your other option is to setup the OpenSER boxes in a truly redundant > configuration using Linux HA (www.linux-ha.org). That way you setup all > your PSTN calls to forward to one shared virtual IP between the boxes. > One > of the boxes is the Master, the other is the Slave. There is a heartbeat > between the boxes that goes at a configurable rate. If the Master fails > then the Slave will take over and it can even be configured for > sub-second > failover. I think there is a article on voip-info.org about this, but > don't have time to look it up. > > Good luck and let us know what you choose to do. > > Ryan > >> All, >> >> I have an Asterisk system that sends PSTN calls to an OpenSER system > to be >> routed. I have a command like this in my extensions.conf: >> >> exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) >> >> There's actually two OpenSER systems for redundancy. I'm trying to > find a >> way to have Asterisk attempt to route the call to one OpenSER system, > and >> if it's down, fallback to another. >> >> Any first thoughts on how to achieve this? >> >> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV > lookups >> are broken. If I issue a series of Dial commands, such as this: >> >> exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) >&g
RE: [Asterisk-Users] Asterisk Dial Failover
I think the Audiocodes boxes run at about $19,000 each. -Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 9:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Doug, We currently are using Digium TE410P boards directly into each Asterisk server. I've been researching various gateways, up to DS3 capacity, to convert PRI to SIP and then allocate the SIP among multiple Asterisk servers. I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and Quintum Tenor CMS ($$). Thanks, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, December 09, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Adam, An Audicodes Mediant 2000 gateway with a couple of PRI's. Why? Doug. -Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Us
RE: [Asterisk-Users] Asterisk Dial Failover
Hi I use a allied telesyn at-vp730 Works quite well ash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: 09 December 2005 16:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Doug, We currently are using Digium TE410P boards directly into each Asterisk server. I've been researching various gateways, up to DS3 capacity, to convert PRI to SIP and then allocate the SIP among multiple Asterisk servers. I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and Quintum Tenor CMS ($$). Thanks, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, December 09, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Adam, An Audicodes Mediant 2000 gateway with a couple of PRI's. Why? Doug. -Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --Bandwidth and Colocation provided by Easynews
RE: [Asterisk-Users] Asterisk Dial Failover
Doug, We currently are using Digium TE410P boards directly into each Asterisk server. I've been researching various gateways, up to DS3 capacity, to convert PRI to SIP and then allocate the SIP among multiple Asterisk servers. I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and Quintum Tenor CMS ($$). Thanks, Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Friday, December 09, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Adam, An Audicodes Mediant 2000 gateway with a couple of PRI's. Why? Doug. -Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transm
RE: [Asterisk-Users] Asterisk Dial Failover
Yes, that's a great question. I'm wondering the same thing. Can these heartbeat apps monitor applications as well as network connectivity? The heartbeat utility at www.linux-ha.org talks about monitoring some standard apps like web servers and such but isn't clear about other apps... like Asterisk or SER. -Original Message- From: John Cianfarani [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 8:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover Ryan/Jonathan, Couple quick questions regarding your setup? Do you operate this in a strictly master/slave setup? Do you have anything(mon/ha's internal status/monitor options) that actually monitors the asterisk process (to determine if it is hung). Or is it only with total box failure to you fail over? Do you use something to sync config/vm/cdr? Rsync/unison? Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --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 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/
RE: [Asterisk-Users] Asterisk Dial Failover
Adam, An Audicodes Mediant 2000 gateway with a couple of PRI's. Why? Doug. -Original Message- From: Adam Robins [mailto:[EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] Asterisk Dial Failover
Ryan/Jonathan, Couple quick questions regarding your setup? Do you operate this in a strictly master/slave setup? Do you have anything(mon/ha's internal status/monitor options) that actually monitors the asterisk process (to determine if it is hung). Or is it only with total box failure to you fail over? Do you use something to sync config/vm/cdr? Rsync/unison? Thanks John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --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 ___ --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] Asterisk Dial Failover
What are you using to terminate the PSTN calls and do the SIP transcoding? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, December 09, 2005 8:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk Dial Failover I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --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 The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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] Asterisk Dial Failover
I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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 ___ --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] Asterisk Dial Failover
Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan > All, > > I have an Asterisk system that sends PSTN calls to an OpenSER system to be > routed. I have a command like this in my extensions.conf: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > > There's actually two OpenSER systems for redundancy. I'm trying to find a > way to have Asterisk attempt to route the call to one OpenSER system, and > if it's down, fallback to another. > > Any first thoughts on how to achieve this? > > I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups > are broken. If I issue a series of Dial commands, such as this: > > exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) > exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) > > ... what seems to happen is that when proxy1 is down, Asterisk waits the > full 20 seconds before returning control. Also, This 20s includes the time > is takes for the other end to answer, so if I put a small value of say 5s > in there, the dial command will probably give up before someone answers at > the other end. Neither is workable. > > Asterisk SHOULD be able to distinguish between a TRYING and no response. > In the event it gets no TRYING response to a dial command within a > specified timeout it should return control and flag an error. If on the > other hand it does get a TRYING response (and maybe a RINGING too) it > should continue to wait until the 20s has expired. > > I can't use dynamic DNS (ie putting two A records for a hostname in DNS) > because Asterisk reads the extensions.conf on startup and also seems to > cache what the host maps to on startup. Subsequent calls to the host > always return the same IP address. > > But... in general... how have people implemented this? > > Help appreciated! > Doug > > > > > > > > ___ > --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 > ___ --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] Asterisk Dial Failover
Hi!
> I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
> lookups are broken. If I issue a series of Dial commands, such as
> this:
>
> exten => 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
> exten => 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)
How about you use ChanIsAvail() before each dial statement? Or try to
Dial with 1 sec timeout or no timeout? Also check ${DIALSTATUS}.
Cheers, Philipp
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