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MediatrixUser

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  1. Dear David, I'm not entirely sure of your setup, but if you are trying the following : Extension 2001 -> SnomOne -> Mediatrix C733 -> PSTN the display number on the line is set by the PSTN provider. You cannot set it on a call basis. You can however control which calls go to which port/line on the Mediatrix, but if you do that the call could fail if the line is in use. There are workaround for this, but unless you dedicate each line to an extension it will not work. If you want to dedicate each fxo/line to an extension, I can probably help with the config of the Mediatrix MediatrixUser
  2. Hello Kaminfeuer, There is two to get the firmware that I know of. 1. Through your Mediatrix product provider. Your reseller should be able to get you the firmware. 2. Directly from the Support Portal. If you need an access I suggest using the forms on those pages : http://www.media5corp.com/en/mediatrix-products http://www.media5corp.com/index2.php?option=com_facileforms&Itemid=197&ff_contentid=144&ff_form=21&ff_applic=plg_facileforms&format=html&ff_frame=1 Also you can tell them their request page is broken : http://wiki.media5corp.com/wiki/index.php?title=Request_form Hope it helps ! If you don't get any answer within 2-3 days, PM me and I can maybe try to reach to them for you.
  3. You are very wewlcome, just happy to help. I installed lots of them and just love the products. Let me know if I can be of any help in the future.
  4. The difference between using the IP address and 50000, is that using the IP address is more like using a trunk. It does not actually use the defined user 50000. In itself it is not a bad thing. If this work for you just leave it just the way it is One downside is that the Qualify (Options) will not work. This means that if the Mediatrix is unreacheable, the calling party will wait for a long time before the call is rejected (32 seconds by default). From the Mediatrix point of vue, the first option (using the user) will always set the Calling Party Number to 50000 (not the name, just the number) because of the PBX (This is not a Mediatrix config). You could use the Call Router and add a Mapping to the Second Route to change the Calling Party Number and replace it with the Calling Party Name (which seems correct). If you want to do this : 1. Create a new Mapping Type Criteria : Calling Name, Transformation : Calling-E164 2. Click the Create Mapping Expression 3. Criteria : .+ Transformation : \0 4. Change back to exten = 999999999,1,Dial(SIP/50000/${EXTEN},80) This should put the Calling Name (which seems correct with 666666666) in the Calling-E164 (This means Calling Number in ISDN) The Call Router seems more like a patch to a somewhat suboptimal config in the VodiaPBX. I'd have to re-install my old VodiaPBX to guide through the config Try the Call Router, it's a seriously underestimated feature of those units.
  5. Lol well of course it is... I guess it was pretty late I just used the User-Agent in the SIP and them got all mixed up with my previous work with Asterisk. An HTTP POST might be possible, but there are some limitation because of the webpage username/password. The Mediatrix unit support DHCP options for auto-configuration. I'll take a look at the options for the future, you are right it might be useful.
  6. Hello Jonitrifulca2, I looked at the trace. What I see before the crash is : -Lots of Registers Your registration timer is set to the default of 120s -> Not a problem You have 4 registrations enabled -> That seems a bit high for such a unit (more on that later) -Lots of Options There is one Options message for each registration, so see registration comments -4 almost simultaneous calls This seems to be a kind of Ring-group in your Asterisk server designed probably to catch the first free BRI port. When the 4 calls hit the Mediatrix unit, two go to BRI and 2 are rejected at the Call Router. One of the two calls that hit BRI is cancelled (the second one). That cancelled call seems to cause the issue. It seems that cancelling twice the call via SIP and then sending a ISDN Connect message causes the crash in the SIP service. I would think it is because the unit is trying from ISDN to cancel a call that no longer exists in SIP. (I can send you a more detailed analysis if you want) That being said I think we can change your configuration to work around the issue. I will need a copy of your configuration (save the modified config as explained in one of my ealier post). From there I can suggest some modification. What I have in mind : 1. Remove the users in the Asterisk server 2. Create a trunk in the Asterisk server 3. Create an outbound route for numbers "5000, 50001, 50002, 5003" and send them to the trunk 4. Remove all registration in the Mediatrix 5. Add a unit registration in the Mediatrix (to go with the trunk config in Asterisk) 6. Change the Call router a bit so that each call goes to all BRI ports. Let me know this would cause an issue with your setup, but I think that if Asterisk sends a single calls that the Mediatrix handles to the first free BRI, you won't hit the problem. Of course your setup can work in future version of DGW v2.0. Upgrading is still possible, but I think that modifying a bit the configuration can save you. Again, if you send me your config I can suggest the modifications required (not on the Asterisk side, tho if you tell me how you configure Asterisk (FreePBX, Config files ?)I can maybe guide you a little) I hope this helps.
  7. Hello jonitrifulca2, I can say that it is what I suspected. It is a crash of one of the Mediatrix service. I believe it is related to SIP. It is a bit late today unfortunately, but I'll take a look at this whenever I have time tomorrow. (It takes a bit of time to analyse a text trace like this, especially since there are lots of calls). As for the default settings, it is an "expected behavior". There is a timer in the unit that starts when the unit boots. If the unit does not boot correctly within that time, it will increase a counter. If this counter gets too high the unit will go back to a previous firmware/configuration. In your case since you don't have any firmware in the second bank (never upgraded) the unit self-factory reset but preserved your config. Can you give me just a little bit more detail on the setup ? I see that multiple calls are sent to the Mediatrix unit at the same time in the trace. I'm curious if you did a trunk in your server ? If so you might consider doing the trunk in the Call Router instead. Just a thought I'll give you more detail when I'm done with the trace.
  8. Hello jonytrifulca, Unfortunately, I can't seem to download the file this time. Maybe upload it as a zip ? As for the download link you found, that are only the MIBS files. This is not a firmware. You will see that inside the zip file files are very different from a firmware. You will unfortunately have to wait for them to send you your access. I'll wait for a new link to download the syslog trace !
  9. Ok, I'm not entirely sure what is the problem yet. We might have to take another log like this one to add information. First I think I see a "End dumping call st" This could be "call stack" if it is a "call stack" this is a serious problem (process crash in the unit). And since I saw "/muse and /libc.so I believe it is a call stack. I also see syslog IDs resetting (from [1E29] to [0000] This usually mean a reboot. I believe the issue happens at the end of a call. First, I suggest setting the LLDP service to manual if you don't use it . This service is used to find VLAN information automatically. This is rarely used. Second, send them a request to access the lastest DGW version. Use the link I posted earlier. Ask for an "access to their portal to download DGW v2.0.27.485" they will send you a document to fill and you can then just send them to get access. You can also ask your reseller, but I think it was much quicker to fill their document for me (got access the same business day). If you don't get access quickly, PM me. I spoke with an engineer of them, maybe he can speed things up. Upgrading might solve this issue. Cross your finger ! Also I saw a fix for b-channel release in their release notes. Worth a try ! Third we can play around a bit with ISDN settings, but since this is on a call disconnect and I don't have complete syslog info (end of line cuts, only last seconds,etc) it is hard do suggest something. Can you extract a text file with your syslog software ? It would give me a bit more information on the call. Summary : 1. Disable LLDP if you don't use it (Take a backup of your config before, just in case (I suggest to go to Management->Configuration Script, click the link at the top "Activate ..." then at the bottom Change "Content" for "Modified Config" and click Export & Download)) 2. Fill and submit the form here : http://www.media5corp.com/index2.php?option=com_facileforms&Itemid=197&ff_contentid=144&ff_form=21&ff_applic=plg_facileforms&format=html&ff_frame=1 asking for DGW v2.0 access. I can also help if this doesn't work, PM me. 3. As long as you wait for DGW v2.0 firmware, try to extract the syslog in a text file so I can have more than the last 5 minutes of logs. Having the completes ISDN messages might give me an idea of settings to change to help. (Like disable unecessary ISDN service, etc.) I hope I can help you see through this !
  10. So that I understand here. When the problem occur, if you go to the service page you see all those services down ? If so, please enable the Diagnostics traces and post the syslog of when the problem occur on this post. I'm very intrigued.
  11. Hello again jonytrifulca, 700-Host Configuration: 85-Current default router is invalid -> If everything works, don't worry about this. A Default Router is only needed if the IP traffic of the device needs to go out of the subnet. 3800-IP Media Synchronization: 110-No master clock is configured => You probably don't use IP Sync so I'd go ahead and disable the IPSync Service (set to manual and stop it in System -> Services) and these errors will stop. About the firmware, because DGW v2.0 uses encryption they make you fill a form before they can give it to you. It's a pretty basic form (name, country, etc) so send a request here : http://www.media5corp.com/index2.php?option=com_facileforms&Itemid=197&ff_contentid=144&ff_form=21&ff_applic=plg_facileforms&format=html&ff_frame=1 and simply ask for a support portal access for the download of DGW 2.0. Right now I don't think you really need to upgrade. I looked at the realease notes and not much is different for ISDN in the lastest versions. (Might still be worth for the future anyway, if you feel like it) About the issue now, if you want to test a setting, go in ISDN -> Basic Rate Interface and set the "Link Establishment" to "Permanent". If your PBX shutdown the link, the unit should re-establish it right away. Prepare to take diagnostic logs if this doesn't work ... Let me know if this help !
  12. Hello jonytrifulca, 700-Host Configuration: 85-Current default router is invalid. -> This error is a network configuration error. It should not prevent anything from working tho. If I were you I would add one manually (I guess you manually set a network IP on the device. Simply go to the Network -> Host page or you can also set one on the interface in Network -> Interface. I usually use the fild in the Host page. 3800-IP Media Synchronization: 110-No master clock is configured. You don't usually need to configure this as the Clock Mode is Auto by default. Look at ISDN -> Status page to see what it is. The IP-Sync is a feature required when you use multiple ISDN device together. Make sure you use a DGW v2.0 recent firmware. Media5 mades lots of fixes for ISDN in the last years. If you have an old version, maybe your issue was fixed. I think the latest is DGW v2.0.27.485. If you have something older than 2.0.24.402 I strongly suggest to upgrade. Especially since the downgrade is SO easy with those units ! I suspect your ISDN provider might put the link to "sleep". Some provider do this to save energy. We could probably find a way to keep the link up if needed. There are additionnal parameters using the CLI/UMN that we can change. If you still have issue. You can post a full Diagnostic Trace and I'll take a look at it.
  13. I don't know much about SnomOne, but I know Mediatrix 3531/3631 are affordable E1/T1 gateays and not so hard to get going. These units also have tons of interop settings, so I'm sure you'll be able to get going.
  14. What kind of ISDN do you need ? BRI/PRI ? Are you looking for something internal or external ? If you need a cheap BRI trunk, I guess a Mediatrix 4401 would do (external 1 BRI port gateway). For a (not-so-cheap) PRI port there is the Mediatrix 3631. Of course internal hardware is usually less expensive than external gateways, but they come with less driver/detection problems and can be easily replaced/expanded without taking the server offline.
  15. Hi YMSL, I can walk you through any Mediatrix product upgrade. First download the new firmware from their web site. (Enterprise SIP 1100/1200) After that extract the content of the ZIP file on your computer let's say c:\Mediatrix You should now have c:\Mediatrix\1204\setup.inf Start PumKIN or another TFTP server and point the root folder to c:\Mediatrix\1204 The Setup.inf is the file the 1204 will be looking for first. Then in the web page : Firmware Download Server Source: Static Firmware Download Protocol: TFTP Username password empty Firmware Download Primary Server Host: IP address of the TFTP server (your PC) Firmware Download Primary Server Port: 69 (default tftp port) Firmware Location Provision Source: Static Firmware Location: Empty Then go the Management -> Admin and select softwareDownload in the system command. Hope it helps !
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