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gotvoip

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Everything posted by gotvoip

  1. gotvoip

    No hang up

    Can you tell us exactly what this means? play 1 250 400 -13 pause 2 250 We are working with the chip manufacturer to be able to set the disconnect tone in different countries. Do you know if the lines there use current disconnect or reverse polarity? Do you know what type of switches are in use? i.e. ericson, siemens? If you route the calls to the auto attendant there is a Hangup Timeout that you can set to say 60 seconds so if all calls are routed to the AA and they don't go to an extension it should hang it up. Not pretty but a workaround until we can figure our the reorder tone or the disconnect detection mechanism.
  2. another thing to keep in mind the new ones are black and have a different processor in them to make the wan ports work and the older ones are white and only have a lan port.
  3. In 1.x we only had ERE and we finally make it simple by putting an * in the pattern with nothing in the replacement and magically all calls would route out that trunk. If you have two trunks you can add 9* to route a call out trunk a and then everything else will go out trunk b. The preference is used as a priority and the lower the preference the higher it is displayed in the dial plan so it is a top down a approach. If you put * as a 10 you will never get to anything below it. If you want to use trunk failover you need to have the same pattern an change the prefernce so the primary trunk will be used first and then when it fails it goes back to the dial plan to find the backup trunk. What are you trying to do in the dial plan that you can't figure out?
  4. sounds like a phone problem. Did you update the phone to the latest firmware? They had a version that came out last month that fixed my one way audio issue. What happens when you make an oubound call?
  5. yes in the settings general there is a Keep CDR Duration (smhd): that defaults to 14d for days. I would set it to 1h for one hour and that should roll the cdr's after that much time.
  6. Is this one of the white CS410's or a the new black ones? If you sent the log level to 9 and turn on all the logging can you attach that to a post?
  7. Hi Kristan, Can we use this as a testimonial on the pbxnsip web site? Thanks!
  8. you need to do this in the dial plan and put something like xxxxxxx 1916* in the replacement so when it sees 7 digits it adds the 1916. that should do it.
  9. I think the all error codes include non sip events like icmp redirects in case the link is really down not just on sip error codes. If the remote side is busy on trunk a it should be busy on trunk b as well so they may not be a good idea. A 486 should be a real busy, the provider should send back a 5xx if they have real problems. Have you hit any issues or are you just trying to think this through?
  10. what happened when you just put 111 in the final stage? That should send it to the exchange server as well.
  11. I think the issues is what happens if you forward the call to an external number and a voice mail system picks up. That would not be to good for a business and it is hard to figure that out easily. What I do instead of using a hunt group is I call my extension and I have a phone in the office and home office registered so they both ring by default and I also have the system setup to ring my cell phone after 5 seconds so that should accomplish what you want. I do get my work vmails on my cell phone when I don't pick up so keep that in mind but the ringing works.
  12. gotvoip

    CDR

    Have you tried the new feature where it sends out a daily report of the CDR's in an email message? It comes in handy to get a snapshot of what happened the day before. There is a pbxnsip CDR Tool product as well that is available that can uses mysql/apache/.php to manage the CDR's. Contact sales@pbxnsip.com for the details.
  13. http://wiki.pbxnsip.com/index.php/CDR_Tool_Setup is the wiki page. It is a good tool to query the CDRs to see the activity on the extensions. For billing you would need to import a A-Z rate table so currently it is good for keeping track of how many total minutes an extension inbound/outbound over a period of time.
  14. Do you know what standard they support? If they support dialogue state it should work but if it is proprietary to Broadsoft or Sylantro it won't.
  15. The name of the setting is "timeout_connected" and you must edit it manually in the pbx.xml core config file (service restart required).
  16. Mine is working fine. I am pretty sure the linux time is independent of the pbxnsip time. It needs to find an ntp server to get the time. Is the dns server working? Can you ping say yahoo.com from the shell. [1] 2007/10/22 21:28:59: Starting up version 2.1.0.2115 [8] 2007/10/22 21:28:59: Route: eth0 d1fb4dd8 fffffff8 [8] 2007/10/22 21:28:59: Route: eth1 01010100 ffffff00
  17. I am assuming it is this setting on the trunk Assume that call comes from user: We were seeing unauthorized messages when calling into exchange so I am also guessing that if you put in extension 101 in there then it will look for account 101's credentials and send that when it is challenged.
  18. sounds like the speech server is using port 80 as well. You can try https and see if you can connect to pbxnsip while they are both running or just move the pbxnsip web interface to 8080 and see if that helps. You can do that from the settings ports and restart the service.
  19. It should work. Did you turn on snmp logging and see what the messages were there? Log SNMP events: YesNo I usually set the log level to 9 and just turn on snmp and see what you get when you poll it.
  20. no you should just have to run the program and copy over the audio prompts. http://wiki.pbxnsip.com/index.php/Installing_in_Linux gives some more info.
  21. In 1.5 if you should have been able to initiate a record if you sent the pbx an info method with record on (snom phone record button) then it would record the call and leave it in the vmail box and you would listen to it like a vmail. If you had the option to forward vmail to emails then it would forward the conversation to your email as a .wav. There should be a record button option so you can enter a * or a # but it can cause problems since it can trigger recording when in a vmail box. In 2.0 the call center edtion was created and that gives you the option to record all calls from an extension, to an extension etc..
  22. gotvoip

    BLF

    Did you try to set up a personal address book for the extension and then use plug and play for the phone? I did that in the past for the 601 and sidecar and it would populate all the buttons. Do you get that far?
  23. Can you set the log level to 7 and attach it?
  24. 2111 is still having RTP issues with the linksys phones.
  25. Can't the supervisor http to their phone and dial it that way? I know the snoms support that.
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