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Steve-Alloy

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About Steve-Alloy

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  1. See attached trace as requested. Calllog.txt
  2. Can someone advise on the best way to achieve the following 0011643XXXXXXX replace with 03XXXXXXX so basically match 001164 and repalce with 0 Any advice is appreciated as this is rather urgent. Thanks.
  3. Thanks for the response but im more concerned about the "timeout" issue. Can you explain why this would occur? Outbound calls through an FXO port disconnect at a 3-5 min internal when there is media for no apparent reason and we see a timeout error. Please explain this. I have a customer who is getting very impatient.
  4. The issue does occur when there is media. If it is a different case, can you elaborate? Can you explain the "timeout" error when there is media? This is occuring to a few guys as posted here.
  5. Can we get a response on this issue? Is starting to occur more and more and we have not had a response. Is this issue being looked into? Will there be a f/w fix that will address this issue? Timeout occurs at random intervals on both inbound and outbound calls. Please advise asap.
  6. Hi, Can anyone elaborate on the below and perhaps give an example of what the below should look like in the invite "The incoming call matches the IP address of the outbound proxy of that trunk and a DID number in the domain of the trunk " referenced from http://kiwi.pbxnsip.com/index.php/Inbound_Calls_on_Trunk I have set this up before using different dial strings like the ones at the bottom of the referenced wiki page however for a large scale deployment (up to 500) what is the best way? Any help is appreciated. Thanks.
  7. I also have a customer with the same issue running f/w 3.4.0.3194, outbound calls only are disconnecting at different call duration's, typically 3-5 minutes. Also getting the timeout error. The f/w version 3.4.0.3201 did resolve this issue but introduced other problems with DTMF detection. Will there be a f/w upgrade/fix for these issues.
  8. Thank you for that after enabling "Requires busy tone detection" (In the PSTN g/w trunk settings) the call is cleared out after approx 5 seconds.
  9. Scenario is very simple to reproduce. Inbound PSTN call routed to AA, if calling party disconnects call at this time the CS410 does not detect the disconnect tone and keeps the line active and in use. Clearing the call out from the web m/ment and disconnecting the PSTN line both obviously resolve the issue but this is not a feasible option. Also occurs if call is terminated at an extensions VM box. Current f/w is 3.4.0.3201 but also occurs on earlier f/w versions. Any advice would be appreciated.
  10. Hi, I have a customer who is having issues adding in a 3rd trunk to his CS410. I have run the license through a decoder which clearly states "Trunks = 3". There is only 2 trunks configured! 1 trunk to an ITSP and 1 to a SIP g/w. Upon trying to add a 3rd trunk the system gives an error stating that trunk limit exceeded. Currently running f/w v.3.3.0.3165, the same error occured on f/w v.3.1.2.3120 & 3.2.0.3143. In my experience with CS410's i have seen inconsistencies with this which can allow more than 3 trunks to be configured without any additional trunk licenses. Anyone else come across this issue? Does the CS410 count the PSTN g/w as a trunk even if it is deleted from the trunks list? Some feedback on this issue asap would be appreciated. Thanks.
  11. Does V.3.X in pbxnsip support Direct SIP URI dialling for outbound and inbound calls? e.g. 30@<IP Address> I really need it for outbound for example: I’m logged into Salesforce and i keep a table of my phones IP address and ext number, in Salesforce can i highlight and dial a number from there. Is the possible to be configured at the PBX level and not on the handset? Please advise in detail as this is fairly urgent.
  12. I have added a static registration with the following "sip:0@127.0.0.1:5062;line=1". Its still not working and all faxes are failing. Does the "sip:0" need to be changed or is that what is needed for a static route to work?
  13. Is there anyway to automatically pass an incoming sip call to a CS410 and have the dial plan configured to pass the call out an FXO port? Reason is i am testing faxing s/w on a Windows PC and have configured a registration in the CS410 with the IP address of the PC which allows the call to come into the CS410 (which i can see from the system logs) but from there the CS410 does not know what to do with the call, i dont want to send it to an extension but rather send it out an FXO port as a T.38 fax. Is this possible?
  14. Nevermind, i missed this setting. Mailbox Explanation Prompt: Always or Not on personal recording
  15. Ive noticed that after recording a personal greeting on the CS410, when a call is answered by an extensions VM box, my personal greeting is played and then the default message of "please leave you message, for further options press the pound key etc..." is also played. Is there anyway to remove this message once a personal greeting has been recorded?
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