intParse (Medisys)

Thanks for your reply. I've solved the problem but forgot to write it here. The previous posts and the wiki link definitely helped. I've followed a similar way as you've suggested; instead of cacert.org root certificate, I've created a personal/test CA cert using the openSSL and copied the Certificate and RSA key to the PBXnSIP page. After that I had to add the CA cert to my "Trusted Root Certification Authorities" via IE. This was just a test to get eyeBeam working with TLS/SRTP. Our client has agreed to get a signed CA Cert from a Certificate Authority (Verisign, Thawtee etc). Thanks again for your help.

OK, I've found this http://wiki.pbxnsip.com/index.php/Getting_...lid_Certificate but I don't think this would help, right?

How do you add the root certificates for pbxnsip? IE complained about certificate being not valid and when I look at the certificate details in the General tab, the validation date shows "Valid from 7/1/2005 to 7/1/2006". How can I extend this validation period for the certificate? ---- Certificate window General Tab --- Certificate Information This certificate cannot be verified up to a trusted certification authority. Issued to: localhost Issued by: Product Development Valid from 7/1/2005 to 7/1/2006 ---- END ---- PBXnSIP is set to propose secure connection. SNOM 360s work just fine. eyeBeam version 1.5.19.2 Build 49847 can register, make calls but can't receive calls. SIP log has the following output; "SIP/2.0 415 No secure channel available for encrypted call" Do you think it is the certificate? if so how can I extend the validation period? Thanks,

Thank you so much. Unfortunately a move to 2.0 is not an option, yet. The project was started in the early days of PBXnSIP and currently the client does not wish to upgrade, although we are tying to convince them. Thanks again.

I am trying to find what params/soap message to send after receiving an ACDAvailable and CallingCardLogin soap message on PBXnSIP v1.5.xx? I have tried to reply with an IVROutput like message but didn't work. Any help greatly appreciated. Thanks. An ACDAvailable soap messsage has these params: {CallID, From, To, Queue, Agent} A CallingCardLogin soap messsage has these params: {CallID, From, To, Account}

Thank you pbxnsip. We've tried and tested the new version on SuSE and no problems, it works. Just wanted to let you know. Best Regards,

Suggestion to Jordan; You might want to use the Auto Attendant and set the redirection timeout value to the recorded message length or more and redirect the caller to where ever you want (the Hunt Group or the IVR node) when the Auto Attendant times out. Would this solve your problem?

Thank you for both, windows and SuSE, updated versions and for taking care of the SRTP problem on Snom phones in a short time. We will try this new SuSE version on our production server and will let you know if something goes wrong but I am sure the problem has been fixed. Thanks again. Best Regards,

I've tested the new version with snom6.5.10 and it works (30 minutes of srtp call and no white noise). Thanks for this update, greatly appreciated. When should we expect to get the SuSE version? P.S. I know it's 4th of July so I am not expecting anything today. Have a safe 4th of July people and easy on the fire works... Cheers,

We are also seeing this problem (Our client using Vegastream 50 as well but I beleive this is not caused by Vegastream). After our tests we've found out that if the recording on the IVR node ends and starts to repeat itself the DTMF tones that are entered during those repetitions are not recognized. At first we didn't notice this because the recordings on the IVR nodes were a bit long so the caller had time to enter the tones and IVR recognized it. This behavior is seen with PBXnSIP v1.5.7, and also with the new version v1.5.10, on either a call from the Vegastream or an internal one. Any idea why this might be happening? Cheers,

Any luck with testing the new version on snom360/6.5.10?

Thanks, hopefully the problem will be solved at the end. Meanwhile, how well do AASTRA phones (new 53i/55i/57i) work with pbxnsip? Any experiences with these phones - especially with SRTP? Looking forward to the new update in couple of days. Cheers,

BTW, I forgot to mention one thing about this release 1.5.2.10; the SRTP voice delay was gone. With the previous version 1.5.2.7 it was close to a 1 sec-delay during the conversations and with this update near to no delay was present. Just wanted to let you know. Cheers,

Thanks for the quick update. As soon as I saw your reply with the new version, I upgraded the windows test environment to 1.5.2.10 and started my tests with snom360 v6.5.10 phones. Unfortunately, on the 14min. mark the white noise made its appearance and deafened us to SRTP hell. The srtp package count was; RTP-RxStat: Dur=859,Pkt=42915,Oct=7553040 RTP-TxStat: Dur=857,Pkt=42907,Oct=7551632

Thanks for your prompt reply. The phones are running snom360 sip-6.5.10, yes it is the version 6.5.10 doing it. There are nearly 300 phones and they are already running on 6.5.10, so, IMHO, downgrading them is not an effective solution. This project is due to go live very shortly. There is a lot of custom developed SOAP based features and language customization we implemented based on 1.5.xxx hence we and the customer very reluctant to upgrade at this stage. We do have plans in place for further development for which will get underway as soon as we launch the current project. We are planning to start the new work with version 2, but for now, we will greatly appreciate if you can provide us the fix for SRTP problem so that pbxnsip 1.5.xxx works with snom360/6.5.10. Our main server is running on SuSe and our test environment is on Windows. So If you could make the fix for Suse and Windows platforms we will greatly appreciate it. (Priority is SuSe though) We are insisting on SRTP because that is one of the main requirements of the project. Although we have tested and used the phones running 6.5.10 successfully w/ srtp turned off, eliminating the srtp is not an option for the current project. BTW, how well do AASTRA phones (new 53i/55i/57i) work with pbxnsip? any experiences - especially with SRTP? Thanks again for your help and looking forward to the PBXnSIP 1.5 fix for Suse (and windows if you could) on your earliest convenience. Best Regards.

Facing a serious problem with Snom360 phones about the white noise being generated on calls after 3-4 minutes (sometimes shorter; 30sec) on an srtp connection. I have tested and seen the same symptoms for srtp calls on both linux(SuSe) and windows (2000) machines. PBXnSIP v1.5.1.7 is running on Suse machine and v1.5.2.7 is running on windows2000 machine. All phones were upgraded to Snom360 6.5.10. I even upgraded them to snom360 7.1.9 version but couldn't get the phones make secure (srtp) calls on v7. (BTW v7 upgrade slows down the snom360 phones. Startup and registration was extremely slow. Keep away from v7 if you own snom360 phones.) I have also tried the sip:xxx.xxx.xxx.xxx:5061;transport=tls line on the outgoing proxy and no luck. In the PBXnSIP 1.5 release notes some of the srtp problems were mentioned as being solved but I don't know if they apply to this white noise situation. I've read some of the threads mentioning this problem but couldn't find any solution to this problem. I am listing the threads I found in PBXnSIP and others forums below (namely asterisk forum thread about the snom white noise); http://forum.pbxnsip.com/index.php?showtopic=93 <-- (Check the last answer or do a text search on "white noise) http://bugs.digium.com/print_bug_page.php?bug_id=5413 <--- (do a text search on "white noise") on the second thread it says "When chan_sip is starting a non crypto call, sending out an INVITE without any sdesc (_SIP_SRTP_SDES is unset), but the reply from the peer includes a sdes proposal, a crypto policy get set up and activated, without the local key being exchanged. At least Snom 5.x always answer INVITE with a a=crypto line and "a=encryption:optional" in the sdp, calling one without setting _SIP_SRTP_SDES gives asterisk using srtp and the snom expecting unenctypted rtp, and nice white noise in the earpiece." So here are my questions: * Is this white noise being generated because of snom neglecting to add the a=crypto line on some of the sip messages while sending them to PBXnSIP? * If it is, is there a work around for this? (w/o upgrading to PBXnSIP 2.0) Any help greatly appreciated, thanks in advance.