[asterisk-users] Grandstream VoIP phones

Sat Sep 22 01:54:07 CDT 2012

Quick update.

Grandstream finally released the first update to theirDP715 firmware,
new v. 1.0.0.8.

Here are the differences:

 1. I can receive calls over secure SIP and RTP
 2. 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: 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 <mailto: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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>
>
>
> --
> _____________________________________________________________________
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> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
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