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About jannies

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  1. If you can give an example of "native" CSTA we can look into it. We do not specialize in CSTA and do not have any knowledge other than what we have done here.
  2. We have changed the HTTP post to the host(domain) and removed the @domain after username and still the same error. We also tried to specify the domain in the MakeCall soap request node with no luck. POST / HTTP/1.1 Content-Type: application/xml Host: pbx.nhb.co.za Content-Length: 324 Expect: 100-continue Connection: Keep-Alive HTTP/1.1 100 Continue <?xml version="1.0" standalone="yes"?><env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://www.pbxnsip.com/soap/pbx"><env:Body><sns:MakeCall><domain></domain>&
  3. That is correct. Our unit test for making a call using CSTA works perfectly on version 60 but fails with the above error on the latest 61.0.2
  4. Hi We have been using the CSTA MakeCall since the last post on this topic in 2012 We are in process of upgrading our Vodia pbx from 60 to 61 and in this functionality is now giving a similar error although we have not changed anything: See the full trace below: POST / HTTP/1.1 Content-Type: application/xml Host: 192.168.x.x Content-Length: 338 Expect: 100-continue Connection: Keep-Alive HTTP/1.1 100 Continue <?xml version="1.0" standalone="yes"?><env:Envelope xmlns:env="http://schemas.xmlsoap.org/soap/envelope/" xmlns:sns="http://www.pbxnsip.com/so
  5. We have been using the CDR URL to post json for quite some time and every now and then the posting stops working. This includes the action URL end points for "When a new call comes in" and "When a call connects to an agent" The log file shows that that requests are pending up and till it reaches 500 where it then starts to drop the webclient requests. [5] 2018/05/03 15:23:58: Action URL request http://xxxapi.xxx.co.za/api/CreateNewCall{ "id":"1326", "from":"\"xxxx\" <sip:8062@pbx.xxx.co.za>", "to":"\"xxx\" <sip:8028@pbx.xxx.co.za>", "callid":"5aeb0d7c222b-n3vc93jb
  6. Unfortunately when i tested the above i had my VPN connection on and only realised this now when trying to test this again. I still think we have a NAT problem but what i do not understand is this do not want to work internally now. When using chrome's WebRTC Internals it seems that it is now trying to use the external IP configured in the IP Routing List when we are coming from an internal IP and this information is coming from googles stun server (stun:stun.l.google.com:19302). See attached image. Any clarity on how this works would be appreciated.
  7. Thank you, we setup the IP Routing List as described in the link and it solved the problem.
  8. We have managed to use the user web dialer to make calls and is working 100% within our LAN network. When performing the same from externalyl the actual call is initiated but no voice or ringing. When we run wireshark on the desktop we can see the that there is STUN traffic trying to connect to the internal IP address of the PBX server which is obviously not reachable from external. Is there any configuration that has to happen so the web dialer knows to use the referring uri or IP instead?
  9. We are receiving JSON cdr posts from our Vodia 56.5 PBX and are experiencing time differences between local time and the other time fields. The only time that is correct is LocalTime. All the timezones are set to CAT+2 including Server timezone, admin timezone and domain timezone. See sample data from the JSON post: "TimeStart":"2017-09-04 12:21:18","TimeConnected":"2017-09-04 12:21:18","TimeEnd":"2017-09-04 12:21:51","LocalTime":"2017-09-04 14:21:18",
  10. Hi We have been testing CSTA MakeCall for some time and all of a sudden we are receiving an error "Invalid header(s): To" from our sip Gateway device. It seems the domain gets appended twice behind the number to be dialed. We have been testing this functionality for weeks and this problem started happening yesterday. Please see partial call log below: [5] 2012/06/06 11:56:21: SIP Tx udp: INVITE sip:0718961395@;user=phone SIP/2.0 Via: SIP/2.0/UDP;branch=z9hG4bK-23ddcf1f2f619a06a96c113862f8dc5b;rport From: <sip:8025@192.168
  11. Thanks very much. This resolved my issue.
  12. Hi I have tested this functionality on a snom ONE free version and it works fine, but when trying to get it working on our production pbx i get the same error as above. Both environments are running version 2011- I have tried the above mentioned suggestions with no luck. Below is the soap conversation of the snom ONE free version that works fine. POST / HTTP/1.1 Content-Type: application/xml Host: Content-Length: 326 Expect: 100-continue <?xml version="1.0" standalone="yes"?> <env:Envelope xmlns:env="http://schemas.
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