[asterisk-users] Grandstream VoIP phones

Fri Aug 31 20:55:00 CDT 2012

Vladimir

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. 

Thanks

Bryant Zimmerman (ZK Tech Inc.)  

----------------------------------------
 From: "Vladimir Mikhelson" <vlad at mikhelson.com>
Sent: Friday, August 31, 2012 9:07 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Subject: Re: [asterisk-users] Grandstream VoIP phones

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: 477744485-5061-8 at BHC.BH.BDH.HB):

[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:*97 at pbx.int.mikhelson.com:5061>;tag=as50c4dc59
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 3 [ 54]: To: 
<sip:471 at pbx.int.mikhelson.com:5061>;tag=436538044
[2012-08-23 23:55:09] DEBUG[14132] chan_sip.c: Header 4 [ 39]: Call-ID: 
477744485-5061-8 at BHC.BH.BDH.HB
[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:471 at 172.17.137.71:5061;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 <vlad at mikhelson.com> 
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 

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