Vladimir I have been working with Grandstream on the DP715 firmware. Can you give me screen shots of your configs or a download of it, and possibly some asterisk config examples of how your system is set so I can try your configs in our test env. We have the DP715 units working with the new firmware. Also are you dealing with US support or other country? I would like to offer your feed back to the US project engineers for the product, and any info such as ticket numbers and support agent names would be helpful. Your experience of thier support is not the Grandstream I know, and I would like to get to the bottom of the issue.
Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 ---------------------------------------- From: "Vladimir Mikhelson" <[email protected]> Sent: Saturday, September 22, 2012 2:55 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[email protected]> Subject: Re: [asterisk-users] Grandstream VoIP phones Quick update. Grandstream finally released the first update to theirDP715 firmware, new v. 1.0.0.8. Here are the differences: I can receive calls over secure SIP and RTP No outgoing calls go through What I observed the phone replies from a different port compared to a port it receives SIP messages on. As a result Asterisk becomes confused. For example, "sip set debug peer 999" would only track messages to the phone. Grandstream's support is beyond the level of criticism. It takes them 10 days to reply to a posted message. It seems their only goal is to close the case. So far I am still to see a single bit of help from them. I will continue updating this thread. -Vladimir On 8/31/2012 8:07 PM, Vladimir Mikhelson wrote: Carlos, So far the experience with DP715 is extremely negative. It all starts with the WEB interface which is only served on port 80, no https, period. There is no login name, just password. The phone worked as expected with insecure SIP and RTP. As I started playing with security the phone started acting up. It randomly took calls, then stopped. It placed calls, then stopped. Following is a sample of a corrupted SIP message Asterisk receives from DP715 (pay attention to Call-ID: [email protected]): [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 1 [ 69]: Via: SIP/2.0/TLS 172.17.137.11:5061;branch=z9hG4bK2f5ce157;rport=5061 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From: <sip:*[email protected]:5061>;tag=as50c4dc59 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To: <sip:[email protected]:5061>;tag=436538044 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID: [email protected] [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]: Contact: <sip:[email protected]:5061;transport=tls> [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]: Supported: replaces, path, timer, eventlist [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]: User-Agent: Grandstream DP715 1.0.0.5 [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 9 [ 80]: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE [2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]: Content-Length: 0 According to RFC 3261, "Call-ID contains a globally unique identifier for this call, generated by the combination of a random string and the softphone's host name or IP address." Interestingly, the problem is intermittent. Some calls go through. Asterisk must be able to process these calls from time to time. Which is strange on its own. On top of everything Grandstream's support organization does not seem to exist for all practical purposes. I opened the case on 08/22/2012. Today, 08/31/2012, I finally received a response, "Sorry for missing your call yesterday. We checked the syslog you sent to us and seems the TLS is shut down. I just got some TLS internal test accounts today and will do a quick test. I'll let you know soon. It took them 9 days to start looking into the issue. I will update this thread with progress. Regards, Vladimir On 8/17/2012 11:30 AM, Carlos Alvarez wrote: On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson <[email protected]> wrote: My primary interest is security. Grandstream claims their intermediate and higher-end models support TLS and SRTP. I am really tired of trying to make Cisco phones to communicate securely with Asterisk. Cisco has a great security model but one has to have their provisioning server for it to function. We've never had customers ask for this, but if doing so is fairly easy we would look at it as just another feature we push. Do let me know how it works out for you. -- Carlos Alvarez TelEvolve 602-889-3003 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
