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Dale

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Everything posted by Dale

  1. Look here for Asterisk for Beagle Bone: http://www.beaglebone-asterisk.org/
  2. I'm using a Snom One Mini running version 4.5.1.1107 Zeta Perseids (snom ONE mini) In the log I see sequences of errors that look like this: [5] 2013/08/01 08:35:23: HTTP 192.168.90.229:3280: Alert(2, 0) [5] 2013/08/01 08:40:40: HTTP 192.168.90.228:4765: Alert(2, 0) [5] 2013/08/01 08:40:41: HTTP 192.168.90.228:4942: Alert(2, 0) [5] 2013/08/01 08:40:42: HTTP 192.168.90.228:4511: Alert(2, 0) They appear in bursts of 6 or more within a few seconds. What do the errors mean?
  3. I did not think to look at this earlier, but I can see from the Service Providers logs and web site that registration is not really being lost. When I looked at this in detail and when I tried calls while the PBX was reporting being in the 408 state, I found the service provider did not think that the registration had been dropped and incoming calls still completed. So you are right, the problem is not as serious as it originally seemed to be. The system will be going into operation soon and the service provider does send an email whenever a call is attempted that fails because the PBX is not registered, so soon I will see how real the problem is. My fall back plan if calls really are dropped is to try to configure the system without registration, where the service provider sends the calls to my static, public IP. Regarding the 100 Trying message, from watching the logs I notice that I only get the 408 email when the 100 Trying message comes immediately (within the same second or two) after the 200 OK. That is, the 100 Trying has gotten out of sequence. It seems that this event confuses this version of the Snom One code (4.5.1.1107).
  4. Thanks for your suggestion regarding DNS. However, even when I enter the IP address I get the same problem. I really think that this is a bug in the SNOM ONE code. Here is why. Even when the web page for the trunk status says "408 Request timeout..." the registration continues to be valid on the VOIP provider and I can successfully complete incoming calls to the trunk. I think the SNOM ONE code is getting confused by the out of sequence "100 Trying" message and declaring the timeout when, in fact, no timeout has occurred from the VIOP provider's perspective.
  5. Thanks. I was able to make it work by setting up DID names on the accounts and leaving the "route to extension" blank in the trunk configuration.
  6. I changed the timeout to 40 seconds. It actually seemed slightly worse with that value rather than the default 30 seconds. I get, on average, about one timeout per trunk per hour. With two trunks that is 48 emails per day....not very practical to leave the email status change warnings turned on. I'm thinking that perhaps I could get around the problem by configuring my VOIP provider to send INVITES to my public IP and not use registration. To do that I need to get the routing for the different numbers working in the PBX.
  7. Thanks. I'll let it run with a keep-alive time of 40 seconds and see how it goes.
  8. Unfortunately, I'm not using 5.1. I'm using a SNOM ONE MINI running 4.5.1.1107. How do I do it with this release?
  9. I think I understand what you are saying, but, being new to SNOM ONE, I don't know how to implement what you suggest. How, in the SNOM ONE gui, do I create the rules in the PBX that will route to different extensions based on the Request-URI or To-header?
  10. I observe that the REGISTER message sent by the PBX seems to occur every keep alive time. When you say to try "putting 40 there" are you suggesting 40 be tried for the proposed registration duration or the keep alive time? When I change the keep alive time to be 40, I see the REGISTER messages every 40 seconds. But my original observation was that the problem occurs only when the "Trying" message gets out of sequence due to a route change affecting the UDP packets. How would the timeout affect out of sequence packet processing?
  11. Is call recording on the Snom One Mini running version 4.5.1.1107 supposed to work? I seem to get no recordings, but no errors either.
  12. It does matter which trunk is selected because one trunk is for the main number that goes to the receptionist and the other is for the back office that goes to a different station. I can change the account names at Vitelity, but, sorry, I'm still a bit fuzzy regarding exactly what you mean by "giving accounts in the domain additional names that match the numbers being called." As you can see the incoming invite and To: show the number that was called, but how do I get the PBX to select the right trunk? What, exactly, would you recommend I put in which field of the trunk configuration to get the PBX to match the right trunk? That is, what rule can I configure to make this work? If you could give a specific example based on the INVITE message above that would be very helpful. Thanks.
  13. I really cannot afford the upgrade price just to fix a bug. My registration duration is set to one hour, but the keep alive time is set to blank which seems to default to 30 seconds. Which of these do you suggest I change to a longer value? What value do you suggest? Is there a way I can get this fixed for my version? This is really killing my customer.
  14. I have a Vitelity Trunk that I use successfully for inbound and outbound traffic. I set both the proxy and the domain to sip29.vitelity.net in the trunk configuration. My problem is that I actually have two separate inbound trunks form Vitelity that I want to route differently in the PBX...that is not working and I've created a separate topic for that.
  15. I've personally found that one way audio on the trunks can be caused by a router that does not properly handle the layer 7 SIP translations if you are behind a firewall. If you can disable the routers attempt to translate the SIP headers you can configure the Snom One Mini to create and process the headers correctly on the Admin Settings Ports page using the "IP Routing List" configuration line.
  16. I have a Snom One Mini running 4.5.1.1107 I observe that about once per hour when the PBX is operating and registered I get email notices saying that the trunk registration has timed out followed by a successful registration email after a 60 second timeout. Sometimes I go for a number of hours without the warning and sometimes I get three warnings within 15 minutes. Upon investigation I found that the trunk did not timeout. I looked carefully at the SIP logs and discovered that this happens whenever the UDP "Trying" message gets out of sequence from the "OK" message. That is, the REGISTER message is sent by the PBX to the SIP ISP, and the replies from the ISP come back first with an "SIP 200 OK" and then with the "SIP 100 TRYING". This seems to happen because of a route change in the internet between my PBX and my SIP ISP. It seems to me that the PBX should not complain about this event since the order of UDP traffic is not guaranteed by the Internet protocol. I think that the PBX should just ignore a "Trying" message that it is not expecting. Is there a way I can fix this so the PBX does not think that for 60 seconds the trunk is not registered? I've run a number of tests on my link and no packet loss is occurring. The emails I get from the PBX are shown below: First: Trunk Vitelity Gateway (2) changed to "408 Request Timeout" (Registration failed, retry after 60 seconds). This is a notification email. Do not reply. Always followed 60 seconds later by: Trunk Vitelity Gateway (2) changed to "200 OK" (Refresh interval 30 seconds). This is a notification email. Do not reply.
  17. I'm having exactly the same problem. I'm using two SIP Registration trunks from the same ISP (Vitelity). Could you please elaborate on what you mean by the "account name of the trunk being what the provider sends when using the trunk"? If my SIP invite looks like below, exactly what field in the trunk configuration should contain exactly what value to make this work? (I've replaced some data with xxx.) The only value that differs when I call different numbers on the ISP is the xxxx431160 portion of the INVITE or the To: line, but putting xxxx431160 as the name for the trunk did not work. INVITE sip:xxxx431160@24.xxx.24.238:5060 SIP/2.0 Via: SIP/2.0/UDP 66.241.99.28:5060;branch=z9hG4bK6684f64f;rport From: "XXXX DALE" <sip:xxxx606310@66.241.99.28>;tag=as60252cef To: <sip:xxxx431160@24.xxx.24.238:5060> Contact: <sip:xxxx606310@66.241.99.28> Call-ID: 552d1e4412f59c7d4d2787c417517bc4@66.241.99.28 CSeq: 102 INVITE User-Agent: packetrino Max-Forwards: 70 Date: Mon, 22 Jul 2013 17:38:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 334
  18. Thanks. I misunderstood the workaround from your earlier reply. Putting a * in the replacement field does work as an acceptable workaround.
  19. And what is the workaround that I'm supposed to use? It appears that I can block no numbers. I need to block 1xxx976xxxx numbers and 411 and 1900 numbers as well as 1xxx555xxxx. How do I do it?
  20. Great! So there will be another point release for version 4.5 for the snom one mini? Any idea how often a bug fix release is produced?
  21. I understand what you are saying, but it does not work. Here are some simple examples. Dialplan 200;-;;[2-9][0-9][0-9]334xxxx;;;false 300;Vitelity Gateway;;*;;;false If I dial on the phone 12313347300 the call goes through. The logfile shows: Dialplan "test": Match 2313347300@pbx.XXXXX.com to sip:2313347300@sip29.vitelity.net;user=phone on trunk Vitelity Gateway Call also goes through if I set the first (#200) dialplan rule to be ^[2-9][0-9][0-9]334.* The call even completes if I set the first (#200) dialplan rule to be 2313347300 As I test more I realize that it even completes if I set the #200 rule to be * This should block all calls, so I must not be understanding something. I have the domain dialplan set to "test" which is the one I'm testing. The extension I'm dialing from has the dialplan set to "Domain Default" Rewrite global numbers is set to "Check domain country code" for the Vitelity Gateway. Why doesn't this work?
  22. When I try ^1[2-9][0-9][0-9]334.* the test area for the dial plan works as expected, but the call goes through the PBX. This is also true when I try 1[2-9][0-9][0-9]334xxxx (Note I'm using 334 here instead of 976 so I can test with a non toll number, the real goal is to use 976 instead of 334.) What is see in the log in either of these cases is: Dialplan "test": Match 2313347380@pbx.XXXXX.com to sip:2313347380@sip29.vitelity.net;user=phone on trunk Vitelity Gateway I also want to be able to block 411 calls. I'm testing with 611 instead. If I have a rule that says 150;-;;1611;;;false and I dial 1611 even though the dial plan test show that rule 150 is matched, the log shows that the call goes through: Dialplan "test": Match 1611@pbx.XXXXX.com to sip:1611@sip29.vitelity.net;user=phone on trunk Vitelity Gateway Why won't the PBX block the above calls? I have the PBX configured with North America 2 digit extensions [2-7]x Country Code: 1 Area Code: 231
  23. I have a couple of issues with configuring a dialplan on a Snom ONE Mini running version 4.5.1.1107. The dialplan does not block calls as expected. The dial plan is simple: 200;-;;^1[2-9][0-9][0-9]976*;;;false 300;Vitelity Gateway;;*;;;false The test area shows that rule 250 matches to block calls to numbers such as 12319761234, but the calls are actually permitted to go through the Gateway when dialed on the PBX. There is no other dialplan assigned. Also, if I run a test number in the test area of the web page for 12319771234 the test results show that the number matches the 250 pattern when, in fact, it does not. 1) Why does my Pattern not block 1xxx976xxxx numbers in the real pbx 2) Why does the test area show that 1xxx977xxxx numbers will be blocked when they don't match the pattern? 3) Is there a different pattern that will actually work to block 1xxx976xxxx numbers?
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