Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
2011/10/5 Adam Moffett adamli...@plexicomm.net ** someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. I just installed 3 Trixbox systems in KVM on Ubuntu. Which GUI or shell did you choose to install and configure KVM ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
hi i have used virtualbox on fedora and installed elastix (like trixbox) . there isnt any problem . have fun On Tue, Oct 4, 2011 at 10:10 PM, Esteban Cacavelos estebancacave...@gmail.com wrote: someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- 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] Asterisk (Trixbox) - VirtualBox - Linux Host
I also installed tirxbox on virtualbox without problems. I guess i have to open a new discussion because the problem is about PCIPASSTHROUGH feature on virtualbox?. I cannot get to work this feature on my virtualbox. There are a few discussions about pcipassthrough but based on KVM or Xen. I haven't found information about implementations on virtualbox. What i am trying to do: Guest system (trixbox) use the digium pci card, using PCIPASSTHROUGH feature. Olivier, i haven't installed KVM yet. If i'll decide to test KVM i will let you know which GUI I choose. Anyone hava tried this feature on Virtualbox ?. 2011/10/11 alireza sadeh seighalan seighal...@gmail.com hi i have used virtualbox on fedora and installed elastix (like trixbox) . there isnt any problem . have fun On Tue, Oct 4, 2011 at 10:10 PM, Esteban Cacavelos estebancacave...@gmail.com wrote: someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- 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 -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
On Tue, Oct 11, 2011 at 9:56 AM, Esteban Cacavelos estebancacave...@gmail.com wrote: I guess i have to open a new discussion because the problem is about PCIPASSTHROUGH feature on virtualbox?. I cannot get to work this feature on my virtualbox. There are a few discussions about pcipassthrough but based on KVM or Xen. I haven't found information about implementations on virtualbox. Have you installed the VirtualBox Extensions Pack[1], which is where the 'experimental' PCI Passthrough support[2] is found in VirtualBox? [1] - https://www.virtualbox.org/wiki/Downloads [2] - http://www.virtualbox.org/manual/ch01.html#intro-installing -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
Yes!, i installed the extension pack. I attached this device to the VM. *03:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface.* But, when i run the VM i got this error. VBoxManage: error: Cannot assign non-shared host interrupt handler: VERR_RESOURCE_BUSY (VERR_RESOURCE_BUSY) VBoxManage: error: Details: code NS_ERROR_FAILURE (0x80004005), component Console, interface IConsole, callee I tried to unbind the pci-stub kernel driver, and then run de VM but without success, because occur the same problem. Those are the pcipassthrough specifications 1. Your motherboard has an IOMMU unit. 2. Your CPU supports the IOMMU. 3. The IOMMU is enabled in the BIOS. 4. The VM must run with VT-x/AMD-V and nested paging enabled. - this is ok. 5. Your Linux kernel was compiled with IOMMU support (including DMA remapping, see CONFIG_DMAR kernel compilation option). The PCI stub driver (CONFIG_PCI_STUB) is required as well. - i am not sure. 6. Your Linux kernel recognizes and uses the IOMMU unit (intel_iommu=on boot option could be needed). Search for DMAR and PCI-DMA in kernel boot log. - i am not sure. For the first three specifications i checked that my hardware support virtualization and VT capability is enabled in the BIOS. Any suggestions. ? 2011/10/11 Warren Selby wcse...@selbytech.com On Tue, Oct 11, 2011 at 9:56 AM, Esteban Cacavelos estebancacave...@gmail.com wrote: I guess i have to open a new discussion because the problem is about PCIPASSTHROUGH feature on virtualbox?. I cannot get to work this feature on my virtualbox. There are a few discussions about pcipassthrough but based on KVM or Xen. I haven't found information about implementations on virtualbox. Have you installed the VirtualBox Extensions Pack[1], which is where the 'experimental' PCI Passthrough support[2] is found in VirtualBox? [1] - https://www.virtualbox.org/wiki/Downloads [2] - http://www.virtualbox.org/manual/ch01.html#intro-installing -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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 -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
I will try some more on Virtualbox and then i'll try on kvm. i'll let you know about the results. 2011/10/5 Adam Moffett adamli...@plexicomm.net ** someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. I just installed 3 Trixbox systems in KVM on Ubuntu. They're emergency PBX's for a few companies who lost their phone systems in a flood. They'll become real machines located on the customer premesis in the near future, but they've been running fine for a couple of weeks as virtual machines. One customer reported gaps in the hold music, but that was the only issue and I have no reason to suspect it's related to being virtual machine. I have not tried VirtualBox. -- _ -- 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 -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
Am 04.10.11 20:40, schrieb Esteban Cacavelos: someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! Hello, We have several Asterisk (not Trixbox) running on OpenVZ but i guess its the same with VirtualBox. The biggest problem if you use something below 1.8 will be that you dont have access to a hardware timing source to get dahdi running, or atleast you will have to do some neat tricks to get this running. You should also think about system ressources cause asterisk will need other ressources than for example a webserver. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- I installed and used Elastix 2.0.3 on VirtualBox 4.x (64bit) but I were constantly troubled by high CPU usage and performance issues. I am not a virtualization / Asterisk expert so I may have missed some aspect of settings or configurations. My general reading on various forums seemed to indicate that VirtualBox is still not the best platform real time application like asterisk. My 2 cents -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. I just installed 3 Trixbox systems in KVM on Ubuntu. They're emergency PBX's for a few companies who lost their phone systems in a flood. They'll become real machines located on the customer premesis in the near future, but they've been running fine for a couple of weeks as virtual machines. One customer reported gaps in the hold music, but that was the only issue and I have no reason to suspect it's related to being virtual machine. I have not tried VirtualBox. -- _ -- 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] Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is installed on a Linux host (Ubuntu server 10.04 specifically). I want to know if it is convenient or not, and the reaseons if i should on shouldn't do it. Thanks in advance.! -- Esteban L. Cacavelos de Amoriza Cel: 0981 220 429 -- _ -- 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] Asterisk / Trixbox 2.6 Streaming MOH Problems
I've tried a number of solutions, but I've been unable to get Asterisk working with streaming MOH without running into the buffer issue. I've tried using various combinations madplay, mpg123, mpg321. I've also tried streamplayer by itself, and in combination with play-fifo ( http://www.freeswitch.org/asterisk_stuff/play-fifo.c ) to try and eliminate the issue. For those that are unaware of the problem, what happens when you use a streaming music source with asterisk is you have a process that is running all the time that pipes MOH into stdout, which is then read by asterisk. When a caller is on hold, asterisk starts reading from stdin, and you get your music on hold. When the caller hangs up, asterisk stops reading from stdin (and the pipe becomes blocking), and a buffer is created (I'm not sure where the buffer resides, although I suspect it's probably the system fifo pipe buffer). The problem becomes, when the next caller comes in, and is put on hold, you will hear that buffer (usually about 20-30 seconds), and then it will jump to the current position in the stream, so you hear an ugly jump between the middle of two songs. There was a magic version of mpg123 that was supposed to solve this problem (0.59r, I believe), but I've been unable to get this to work. For those interested, I'm streaming music off of a Barix Instreamer, attached to a satellite radio source (and yes, I'm paying the proper licence fees). The only thing I've found that works so far is a pretty ugly (although ingenious) hack as seen here (http://www.mail-archive.com/asterisk-users@lists.digium.com/msg197299.html), which creates its own host of problems (such as not being able to do a restart when convienent since it generates a call on its own (that's always running), so it's never convienent for asterisk to restart. Also, when I do restart asterisk, I have to restart the call, so I'd prefer having to go this route if at all possible. Another solution would be if asterisk could spawn a new process for every MOH caller. Is this possible? Does anyone have a successful deployment of streaming music on hold that they'd care to share? I'm using Asterisk 1.4 as part of Trixbox 2.6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
How many accounts do you have? If just one, then a single peer should be fine but they should be sending the destination exten as a DID, obviously they are not. I think the burden of fixing it lies with them? What carrier is this? On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett asterisk-us...@ics-il.net wrote: I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Do you know enough about Trixbox to tell me where they need to fix their misconfiguration, or is it a Trixbox design flaw? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Possibly if I could take a look at their GUI and custom contexts. That could be quite a bit of work On Tue, Feb 10, 2009 at 10:26 AM, Mike Hammett asterisk-us...@ics-il.net wrote: Do you know enough about Trixbox to tell me where they need to fix their misconfiguration, or is it a Trixbox design flaw? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 8:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Yes, they should fix this on their side, otherwise DID routing will not work. If you don't need it, you just need to create a DID entry for any/all or any/any, I cannot remember which it is right now, but it should be apparent when you look at it. The s extension is only used when no DID or extension is received. Thanks, Steve Totaro On Tue, Feb 10, 2009 at 9:49 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They must have changed something after I complained because it no longer references the incorrect phone number. I did disable However, it still wants to send everything to the s extension. Everything I have worked with before has sent calls the the DID's extension (a call to 888777 goes to exten = 888777,1,blah). Is this something they can change in Trixbox? http://pastebin.com/fa8b4f4e I highlighted the lines that contain the s extension. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
It's a local CLEC, Essex Telcom. The burden does lie with them, but I doubt they'll fix it since if you provision a grandstream, it works just fine. I have a total of 5 numbers with them. Two are on the server that is experiencing issues. Another is on a different server with no issues. The remaining two aren't provisioned anywhere. I'm going to be adding another number shortly. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@totarotechnologies.com Sent: Tuesday, February 10, 2009 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox How many accounts do you have? If just one, then a single peer should be fine but they should be sending the destination exten as a DID, obviously they are not. I think the burden of fixing it lies with them? What carrier is this? On Tue, Feb 10, 2009 at 10:09 AM, Mike Hammett asterisk-us...@ics-il.net wrote: I disabled that last number's registration and moved to a new number (to test each number individually without the sip debugging from the others). I waited maybe 5 minutes and I restarted Asterisk to ensure the other side was done with whatever it was doing. I called the second number (8152641125) and the first number (8159911010) shows up as the peer. Not only that, but with this number, there's no compatible codecs. I ensured that both entries in sip.conf were the same other than things that needed to be different such as username. I even had that entry have allow=all. I still get the codec error. http://pastebin.com/f5b826d62 I highlighted the lines of interest. 34 is the peer issue whereas 42 is the codec issue. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@first-notification.com Sent: Tuesday, February 10, 2009 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Mike, Please explain the problem more clearly and post a pastebin that shows the problem and only the problem, not a huge SIP dump. If you could point out the line numbers where you suspect an issue. Thanks, Steve On Mon, Feb 9, 2009 at 10:56 PM, Mike Hammett asterisk-us...@ics-il.net wrote: Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Can anyone help me determine where the problem lies and how to fix it? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com From: Mike Hammett Sent: Thursday, January 15, 2009 1:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk - Trixbox My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Your carrier is running Trixbox? That is scary. Anyways, they are obviously routing calls to the wrong machine. If your side worked properly before and now does not, then they have to explain why. That would be my stance anyways. Thanks, Steve On Mon, Feb 2, 2009 at 10:18 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
I don't think scary is a strong enough wordterrifying? horrifying? abominable? PaulH Steve Totaro wrote: Your carrier is running Trixbox? That is scary. Anyways, they are obviously routing calls to the wrong machine. If your side worked properly before and now does not, then they have to explain why. That would be my stance anyways. Thanks, Steve On Mon, Feb 2, 2009 at 10:18 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Yeah. They were running a Clarent switch and that's the one that came down. They also had\have a Coppercom switch. The Clarent was old, though I really didn't have any problems with it. I could never get the Coppercom to work with Asterisk (though I'm an expert at neither) and their tech support told my carrier to fly a kite when we were having T38 issues. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Steve Totaro stot...@totarotechnologies.com Sent: Monday, February 02, 2009 9:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Your carrier is running Trixbox? That is scary. Anyways, they are obviously routing calls to the wrong machine. If your side worked properly before and now does not, then they have to explain why. That would be my stance anyways. Thanks, Steve On Mon, Feb 2, 2009 at 10:18 AM, Mike Hammett asterisk-us...@ics-il.net wrote: They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
They are running Trixbox 2.6.1.10 and I'm running Asterisk 1.2.12.1. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Mike Hammett asterisk-us...@ics-il.net Sent: Thursday, January 29, 2009 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
Should Trixbox be sending calls to the s extension in the first place? I can't set an s extension because there are many independent phone numbers in that context that worked fine before my provider switched to Trixbox. Also, why would the 8159093011 phone number be showing up in the sip debugging when that number isn't even present on that machine? - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -- From: Adrià Vidal adriavi...@gmail.com Sent: Friday, January 16, 2009 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Trixbox On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Trixbox
On Thu, Jan 15, 2009 at 8:00 PM, Mike Hammett asterisk-us...@ics-il.net wrote: My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com I think you need something inside [DID-incoming] like for example... exten = s,1,NoOP(-incoming call---) exten = s,n,Playback(wellcome) # Looking for s in DID-incoming (domain 208.100.1.33) # Reliably Transmitting (no NAT) to 208.1.87.235:5060: # SIP/2.0 404 Not Found -- -- Adrià Vidal adriavi...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Trixbox
My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I solved the s@ problem on the other server by adding insecure settings, but that didn't seem to solve it on this one. http://pastebin.com/f5151341f - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
