<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">Vladimir<br />
<br />
DP715 Phone Only. <br />
<br />
Also another point of note. At present I would not promote the DP715 as an executive level advanced feature phone at best it is a residential grade unit with the current firmware. This last firmware release fixed some major issues but crippled the unit from four concurrent calls to two if you are using g729. This is a big kick in the paints and shows some possible engineering shortcomings of the units. We are talking to the engineers to see what their product will look like once the firmware is closer to production ready. At current we have downgraded our release state from production to beta on our network. Several customers are very please with the units but it has failed to meet the expectations of others. <br />
<br />
Take a close look at the release notes for the DP715 this will infer some of what was wrong in the first release and give you a kind of idea of where the product still needs to go. <br />
<br />
<div>Thanks<br />
<br />
Bryant Zimmerman (ZK Tech Inc.)<br />
616-855-1030 Ext. 2003</div>
<br />
<br />
<span style="font-family: tahoma,arial,sans-serif; font-size: 10pt;"><hr width="100%" size="2" align="center" />
<b>From</b>: "Vladimir Mikhelson" <vlad@mikhelson.com><br />
<b>Sent</b>: Saturday, September 22, 2012 2:55 AM<br />
<b>To</b>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<b>Subject</b>: Re: [asterisk-users] Grandstream VoIP phones</span><br />
<br />
Quick update.<br />
<br />
Grandstream finally released the first update to theirDP715 firmware, new v. 1.0.0.8.<br />
<br />
Here are the differences:<br />
<ol>
<li>I can receive calls over secure SIP and RTP </li>
<li>No outgoing calls go through </li>
</ol>
<p>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.</p>
<p>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.</p>
<p>I will continue updating this thread.</p>
<p>-Vladimir<br />
<br />
</p>
<p></p>
<br />
<br />
<div class="moz-cite-prefix">On 8/31/2012 8:07 PM, Vladimir Mikhelson wrote:<br />
</div>
<blockquote cite="mid:50415FB9.3000806@mikhelson.com" type="cite">Carlos,<br />
<br />
So far the experience with DP715 is extremely negative.<br />
<br />
It all starts with the WEB interface which is only served on port 80, no https, period. There is no login name, just password.<br />
<br />
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.<br />
<br />
Following is a sample of a corrupted SIP message Asterisk receives from DP715 (pay attention to Call-ID: <a href="mailto:477744485-5061-8@BHC.BH.BDH.HB" class="moz-txt-link-abbreviated" moz-do-not-send="true">477744485-5061-8@BHC.BH.BDH.HB</a>):<br />
<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 0 [ 14]: SIP/2.0 200 OK<br />
[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<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 2 [ 57]: From: <a href="mailto:sip:*97@pbx.int.mikhelson.com:5061" class="moz-txt-link-rfc2396E" moz-do-not-send="true"><sip:*97@pbx.int.mikhelson.com:5061></a>;tag=as50c4dc59<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To: <a href="mailto:sip:471@pbx.int.mikhelson.com:5061" class="moz-txt-link-rfc2396E" moz-do-not-send="true"><sip:471@pbx.int.mikhelson.com:5061></a>;tag=436538044<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID: <a href="mailto:477744485-5061-8@BHC.BH.BDH.HB" class="moz-txt-link-abbreviated" moz-do-not-send="true">477744485-5061-8@BHC.BH.BDH.HB</a><br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 5 [ 13]: CSeq: 102 BYE<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 6 [ 51]: Contact: <a href="mailto:sip:471@172.17.137.71:5061;transport=tls" class="moz-txt-link-rfc2396E" moz-do-not-send="true"><sip:471@172.17.137.71:5061;transport=tls></a><br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 7 [ 43]: Supported: replaces, path, timer, eventlist<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 8 [ 37]: User-Agent: Grandstream DP715 1.0.0.5<br />
[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<br />
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 10 [ 17]: Content-Length: 0<br />
<br />
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."<br />
<br />
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.<br />
<br />
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.<br />
<br />
I will update this thread with progress.<br />
<br />
Regards,<br />
Vladimir<br />
<br />
<br />
<br />
<div class="moz-cite-prefix">On 8/17/2012 11:30 AM, Carlos Alvarez wrote:<br />
</div>
<blockquote cite="mid:CAFn1dUFmTSPnwydAAVHSpifh=be9TYVwDat_+NHcvQ7-cYRASw@mail.gmail.com" type="cite">
<div class="gmail_quote">On Fri, Aug 17, 2012 at 9:08 AM, Vladimir Mikhelson <span dir="ltr"><<a href="mailto:vlad@mikhelson.com" target="_blank" moz-do-not-send="true">vlad@mikhelson.com</a>></span> wrote:<br />
<blockquote class="gmail_quote" style="border-left: #ccc 1px solid; margin: 0px 0px 0px 0.8ex; padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF">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.<br />
</div>
</blockquote>
<div><br />
</div>
<div>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.</div>
<div><br />
</div>
</div>
-- <br />
<div>Carlos Alvarez</div>
<div>TelEvolve</div>
<div>602-889-3003</div>
<div><br />
</div>
<br />
<br />
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