<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">Vladimir<br />
<br />
We are testing the DP715 very aggressively. We have been please with the units for the most part, but we too have been working bugs with Grandstream. We have several in so far and a number of feature requests as well. I deal directly with several of the support engineers and they bring in the developers when necessary. I would be open to working with you on your issue. If I can create validation tests for your items and reproduce the issue I have had great success getting them to take note and address issues they really do want to address issues. In less than two weeks they have given me test builds address two of our issues and they are working on several others. Because of the cooperation of Grandsteam we are close to being able to offer the DP715 phones to our customers. Even then they will have more items to address to allow for full feature deployments but they are serious about the DP715 product. <br />
<br />
<div>Thanks<br />
<br />
Bryant Zimmerman (ZK Tech Inc.)
</div>
<br />
<span style="font-size: 10pt; font-family: tahoma,arial,sans-serif;"><hr width="100%" size="2" align="center" />
<strong>From</strong>: "Vladimir Mikhelson" <vlad@mikhelson.com><br />
<strong>Sent</strong>: Friday, August 31, 2012 9:07 PM<br />
<strong>To</strong>: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com><br />
<strong>Subject</strong>: Re: [asterisk-users] Grandstream VoIP phones</span><br />
<br />
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">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"><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"><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">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"><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 type="cite" cite="mid:CAFn1dUFmTSPnwydAAVHSpifh=be9TYVwDat_+NHcvQ7-cYRASw@mail.gmail.com">
<div class="gmail_quote">On Fri, Aug 17, 2012 at 9:08 AM, Vladimir
Mikhelson <span dir="ltr"><<a target="_blank" href="mailto:vlad@mikhelson.com" moz-do-not-send="true">vlad@mikhelson.com</a>></span>
wrote:<br />
<blockquote style="margin: 0px 0px 0px 0.8ex; border-left: 1px solid #cccccc; padding-left: 1ex;" class="gmail_quote">
<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 />
<fieldset class="mimeAttachmentHeader"></fieldset>
<br />
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