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Kaj.Noppen

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  1. *BUMP* I'm still looking for this. NexSIP, pbxnsip, you guys still here? Thanks!
  2. Yep, tried that. We only have 1 trunk configured. What could the inbound / outbound trunk add to the picture? It doesn't change all that much. Fact will remain that it sends out our SIP account name, instead of a number. See codebox below, this is the invite the PBX makes when it starts a new call for the redirection. Real: INVITE sip:316148xxxxx@test.winitu.com SIP/2.0 Via: SIP/2.0/UDP 10.1.2.250:5060;branch=z9hG4bK-0cef403e1f3262153d574353882d0166;rport From: "ocsproduction" <sip:ocsproduction@test.provider.com>;tag=852891320 To: <sip:316148xxxxxx@test.provider.com;user=phone> Call-ID: 9415cdaa@pbx CSeq: 2199 INVITE Max-Forwards: 70 Contact: <sip:ocsproduction@10.1.2.250:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.0.1.3438 Related-Call-ID: 1761cef4@pbx Content-Type: application/sdp Content-Length: 233 Instead of the sip:ocsproduction@test.provider.com, we want it to give the extension ANI, which is a DID. So e.g. FROM: <sip:0123456DID@test.provider.com>.
  3. This does not work. We have one SIP account, which has multiple DID numbers attached to it. When we set the ANI on the extension it works correctly when calling from the extension. However, when calls are forwarded / redirected, it does NOT use the extension ANI, but the SIP account name. This is not expected behaviour, well.. at least not wanted in our case. All that we would like is some kind of a setting where we can choose how the FROM header is generated for redirection..
  4. Hi, sorry to drag up such an old topic, but this is exactly what we have been trying to do for some time now. Could you elaborate on how one could achieve this? I have been trying to loop the call through the PBX to get rid of the original caller, but I cannot get it to work. Is there any other option since this post? Is this still possible by looping? Duplicating the trunk (as nexSIP suggests) is not really an option, since we want to do this for ~ 10 different DID's. This would mean we would have to create 10 trunks, but our SIP provider only allows 2 registrations per account. Thanks!
  5. Could someone with acces to the wiki / support site please update this, took me some good time to find this solution... If it was put on the wiki correctly, the php script on the server work almost directly. Also, if this person could make it possible to download the files, in stead of copying them, it would be a lot more easy. Also, the php on the wiki is malformed and has some small errors.
  6. At the moment I have tried editing the settings, renewing the certs and creating the config on a windows and linux machine. However I am still unable to get it to work. I'm trying to get in touch with an SSL / windows cert expert, but no luck so far
  7. This is for OCS - non R2 We're working with R2. 1. I guess this is a question towards pbxnsip, I have no clue. Sounds plausible. But isn't crl the base of secure communications? 2. I could give it a try, but it doesn't state the R2. I would image that MS has included the hotfix in R2. Anyways, I'll try to do some more magic with the certificates. Perhaps I'm missing a step somewhere. I'll try to get a nice wireshark trace as well. Thanks for the help! Kaj Edit: Is it possible to somehow alter the port to which pbxnsip sends the TLS traffic to? It looks like it is sending it to port 5061, but this is the wrong port, since OCS wants internal TLS traffic on 5061, external (whatever type of transport) on port 5060. The audiocodes manual Jan referred to also states that they change the TLS port to 5060. Also: The cert has to be valid, since when I go to https://url-of-pbxnsip/ on the mediation server it says the cert is valid.
  8. I did do that. It looks like the pbxnsip does not accept the certificate presented by the Mediation server. So it is possible? Have you had any experience doing this?
  9. Hello everybody, At the moment we have setup a nice testing environment with OCS R2, integrated with Exchange UM. Everything was working fine with TCP transport. However we want to step up the security a notch and are trying to get TLS transport working. However, I seem to have gotten the settings correct for TLS, but calling to OCS does not work. Calling from OCS is working fine. Calling to OCS (from PSTN for example) however is not. As far as I can see the TLS handshake fails, which is odd, because it does work the other way. What I am wondering, is there anyone else who has been able to set up pbxnsip to OCS with TLS? If that's the case, we'll be trying to get some real certs instead of the one's from our own enterprise-CA. Thanks! Kaj
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